@ChrisAdmin wrote:

Hi there friends. I need to implement some features to freePBX.
Also can this be achieved with multicast for using more than 10 extensions groups?

1. Distribute music to all or a group of extensions registered in freePBX. Source: any online streaming address, spotify, etc. according to the limitations of the case.
2. Make a call or paging call to all or a group of extensions registered in freePBX.
   It should interrupt the functional music mentioned above.
3. Call an extension “X” and triggered/send prerecorded messages to all or a group of extensions registered in freePBX. Source: Audio files in wav or Mp3 format.

Thank you all, Chris.

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@robertoa26 wrote:

I’ve installed two separate FreePBX machines and I"ve linked them together using a trunk. I can call Ext. to Ext. between the two just fine. Everything is set up correctly. My problem is, I don’t know how to set them up so that I can park on an extension on PBX1 and pick that parked call up from a phone on PBX2 and visa versa.

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@claloano wrote:

I was updating Framework FreePBX15.0.16.22
And here is this error:

Please wait while module actions are performed
Downloading and Installing framework
Downloading framework 16079046 of 16079076 (100%)
Error (s) downloading framework:
File Integrity failed for /var/www/html/admin/modules/_cache/framework-15.0.16.53.tgz.gpg - aborting (GPG Verify File check failed)
Updating Hooks … Done

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@Cam wrote:

As far as I can tell the blacklist module has received virtually no love or attention practically since its inception. It could be a much better tool in the fight against spam callers if it were brought up to date.

The two main improvements I would suggest are blocking based on Caller ID name, and partial number matching. Caller ID name is easy to explain - rather than looking at the Caller ID number you look at the received Caller ID name and if it matches a particular string it gets blocked.

Partial number matching would let you say that if a caller ID number begins with the specified digits it would be blocked. Now you may wonder how useful this would be, but consider this.

If you go to https://www.telcodata.us/new-exchanges-report and pick a recent month to display, it will show you all the new thousands blocks of telephone numbers assigned in that month. While it is understandable that big providers like Verizon or Comcast might continually need new number block assignments as they add customers, it’s a bit more puzzling why some companies you have probably never heard of would need so many new numbers, until you realize exactly how those numbers are often used.

As least two of the less well known company names on those lists are companies that I recognize as being the originators of better than 90% of the spam calls that come into my PBX. I’m not going to name them because they might take offense to being called out as spam enablers, and they probably have lawyers. But a few months back, I simply started blocking all calls that come from their exchanges in my home state and surrounding states, plus a couple other states that tend to be big originators of spam calls, and that has dramatically cut the number of spam calls that make it through the system. Unfortunately, it seems that many phone spammers and con artists don’t like to hold onto the same numbers too long, because people start to recognize and block those numbers. And these spam-enabling telcos just keep giving them new numbers to annoy us from.

If there were a requirement that phone companies had to prove that they were actually adding new customers, and not just allowing existing customers to keep switching numbers to avoid detection and blocking prior to being assigned any more new thousands blocks, it would be much easier to curb the flow of spam calls. But in the meantime the best way I have found to block spam calls it to block all calls from thousands blocks assigned to those two companies. This means blocking based on the first seven digits of a ten digit number. Right now I’m having to do it using a very expansive block of dialplan in extensions_override_freepbx.conf, which is obviously far from an ideal way to do it. But it also means that when new thousands blocks are added for those companies I need to go in and modify my custom dialplan, which is a real pain in the butt.

Which bring me back to my idea that if you could specify that a number in the blacklist only had to be a partial match for the first x digits of a number that is y digits long or longer - in this case x would equal 7 and y would equal 10, but it might be different in other countries - it would be easier to block calls originating from these spam-enabling telcos. If anyone else thinks this is a good idea, maybe the FreePBX developers might want to consider giving the blacklist module a bit of an upgrade.

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@vladvlad12 wrote:

So I just installed the freepbx and registered 2 cisco cp-8845 phones.

The call is working, the jitter on audio is at max 3, but on video the jitter is over 1000, with a max of 3000+. What can cause this problem?

I also have a SIP trunk with a CUCM 11.5 and if I call on a phone in the CUCM it works perfectly.

Also I wanted to ask if there are some setting that allow phones to communicate phone-to-phone after the link is estabilished, instead of phone-to-pbx-to-phone

ATM my freepbx is on a server, and have allocated 8GB RAM.

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@posi211 wrote:

A while back someone talked about some code they created to take the email from Fai2Ban when it bands someone it adds the IP to the firewall blacklist.

Anyone here willing to share?

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@mikelandon wrote:

Hi Guys,

I have been trying to find a way to receive FAX over IP on the Freepbx for days, but no luck, could anyone please help me.
My configuration is quite simple, I have a vodafone sip trunk line, the voice part works well.
I have a brother MFC9332 printer connected to the network, I will using that to receive fax.
Please note, I dont need to send fax, I only need to receive fax.

Could any advise on how to setup fax on the freepbx? I would be forever grateful if someone can show me step by step instructions to setup the fax function.

Thank you!

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@johntgs wrote:

how can I turn that off? Under notification I removed my email for intrusion detection? Must I disable it from a command line / file location? Is there check box someplace. I get dozens a day so I like that it works. This started after I rebooted the server the other day.

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@NeoMod wrote:

I was trying to configure the GoogleContacts function inside CID Superfecta, but I’ve encountered an error with the authentication process with my Google Account.

This app has not yet been verified by Google and therefore cannot use the Sign in with Google feature.

I have also found out aninteresting topic where the same error is mentioned and the user @jamesg224 reports a successful authentication with his google account where the 2FA is enabled.

I have tried - as per usual - generating an “app-specific password” but still the authentication fails with the aforementioned error.

Also, in the same topic mentioned above, I found a possible explanation for “enabling untrusted apps” but that only applies to GSuite Admin accounts.

If possible, I would like to know how to correctly connect CID Superfecta with a Google Account (not a GSuite Account) where the 2FA is enabled.

Thanks in advance for your help.

(while trying to do so, I have also verified that the information reported here is incorrect for CID Superfecta v.15.0.2.23; trying to reset ClientID and ClientSecret to those values lead to an “app not found” error from Google at the sign-in/auth stage).

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@Keithc wrote:

I am looking to create, run and copy over backups via ansible from multiple systems.

Is it possible to create a backup from the command line? Is it possible to run a backup from the command line?

Is there anywhere I can see examples of this? I have tried googling this but not coming up with anything definitive.

Running the following

/var/lib/asterisk/bin/backup.php
backup.php

options:
–id= - a valid backup id
–astdb=<restore|dump> - dump or restore the astdb
–data= a serilialized string of the astdb dumb to restore.
Can also point to a file contianing the serializes string

/var/lib/asterisk/bin/backup.php --astdb=dump
Array

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@aboggs wrote:

We have a mix of Polycom VVX411’s and Cisco SPA504 IP phones.

I would like to create 2 parking bins on each phone.
Press the button once to park a call on that buttons. Then have that buttons show busy when a call is parked. Then press the button on another phone to retrieve that parked call.

Any help would be appreciated.
-Adam

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@JesseKnox wrote:

I’ve been pulling my hair out trying to get a remote extension setup but I can’t get it to register for the life of me. I’m a network engineer but don’t really have a deep understanding of how SIP works.

Anyways, I have a FreePBX server at my office and a Grandstream handset at my house. I followed the Freepbx Wiki guide when trying to set up the remote extension but it fails to register. All handsets local to the PBX work flawlessly.

Running a tcpdump on the pbx, I can see the traffic hitting the pbx but it doesn’t appear that it is sending any traffic back which I believe is the problem. Here’s what I’m seeing…

11:08:39.418000 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0
11:08:43.421929 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0
11:08:47.426417 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0
11:08:51.430768 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0
11:08:55.432909 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0
11:08:59.433804 IP wsip-X-X-X-X.21061 > 192.168.10.252.sip: SIP: REGISTER sip:X.X.X.X SIP/2.0

The first address is the WAN IP of my house and the address after the “REGISTER” is the WAN IP of my office.

I currently have NAT set to “yes” in the extension settings, I also have NAT set to “yes” in SIP settings and have my external address and local networks all setup correctly.

On the remote Grandstream phone, I have tried all of the different NAT settings but they all fail to work.

I have googled for many many hours and have not found a solution. Any help would be greatly appreciated.

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@freepbx_is_free wrote:

I am having an issue with inbound and outbound calling on a brand new FreePBX 15.0.16.53 (Current Asterisk Version:16.9.0) installation.

1. I have configured the FreePBX server and registered a phone at extension 101.
2. I have created both a pjsip and chansip connection to my Nextiva trunk and they both say that they are registered (I know both are not needed).
3. I created an inbound route directly to extension 101 on a registered phone.
4. I created an outbound route using the pjsip trunk connection because I think that is what I should be using, (ideally?).

See ->

PJSIP

Nextiva_1/sip:myhost:5060 Nextiva_1 Registered
Nextiva/sip:myhost:5060 Nextiva Registered
Nextiva2/sip:myhost:5060 Nextiva2 Registered
Nextiva3/sip:myhost:5060 Nextiva3 Registered
Nextiva4/sip:myhost:5060 Nextiva4 Registered

Objects found: 5

SIP

Host dnsmgr Username Refresh State Reg.Time
sip:myhost:5060:5060 Y 123456789 3585 Registered Thu, 04 Jun 2020 18:55:53
1 SIP registrations.

CHANSIP

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
Nextiva_ChanSIP/123456789 208.73.146.93 Yes Yes 5060 Unmonitored
Nextiva_ChanSIP2/123456789 208.73.146.93 Yes Yes 5060 OK (63 ms)
2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline]

I have a lot of trunk registered because I was trying to find the right configuration.

When I call my DID I get a message from Nextiva saying the wireless caller is not available. This is the same message I get when no SIP trunk is registered.

I have disabled the firewall on the FreePBX server. Also, we have a Sonicwall in front of the PBX that I increased the UDP timeouts on to 120. I installed Zoiper on a machine on the same LAN as FreePBX and can connect to the Nextiva SIP trunk and make and receive calls.

This is the format I am using for my registration string.
[Authentication Name]:[Authentication Key]@[Nextiva Host]/[Authentication Name]

Can anyone point me in the right direction to get these inbound and outbound calls functioning?

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@tonyg wrote:

All,
I know that this is a hot topic and I have read through many of the previous threads. My question is a bit more specific.
I have customers that are considering keeping their employee’s home permanently. They want to provide business phones to them. They want to use a softphone but I am aware of the kari’s law issues.
My question, is…given they are home and have at least one phone (if not 2), do we still need to provide 911 on the softphone? If the company instructs the user NOT to use the softphone for 911, does that give us a pass?
how are others handling this?

thanks

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@tjthorson wrote:

Friends,
I have been beating my head against the wall all day on this and just cannot figure it out. I am setting up a FreePBX distro with a SIP trunk from FirstComm. They claim there is no ID or secret to use, they use the external IP address for security. On my PBX, I have ETH0 as the LAN and default gateway there. ETH1 is the external IP that Firstcomm wants me to use. I have a static route for the SIP trunk IP out through ETH1. Firewall is turned off. This is the same type of config I have done for years on many different providers. I have been playing with trunk settings, adding and removing a myriad of settings and cannot get calls to move in or out. If I do a SIP debug (when I set qualify=yes), I see the PBX attempting to register and not getting a reply.

Retransmitting #1 (NAT) to 216.159.230.***:5060:
OPTIONS sip:216.159.230.*** SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK31576365;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.1.11;tag=as7834a30a
To: sip:216.159.230.***
Contact: sip:Unknown@192.168.1.11:5060
Call-ID: 4e2942ee10bc8e6b18961ced3baea28c@192.168.1.11:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.53(16.9.0)
Date: Fri, 05 Jun 2020 00:26:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

If I try to make a call in our out - the call just sits and never goes anywhere, just silence. The SIP provider has no clue… Im thinking that I need to put the external IP into that string its sending (instead of the ETH0 local lan .11 address), but I don’t know where to do that? Ive never had to do this before with any other provider… WHat can I send? The current trunk settings are:

type=friend
host=216.159.230.***
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
qualify=yes

Currently inbound is blank - but I have tried many of the above settings in there too…If I ping the SIP trunk 216 address from the console, it responds fine. Traceroute shows it going out through ETH1 so I think the networking part is OK

Hopefully this makes sense - let me know what else I can give. This is the latest FreePBX distro with all updates as of 6-4-2020

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@Lutiana wrote:

Hello everyone!

FreePBX: 15.0.16.53
Asterisk: 16.9.0
Cisco 7975G w/ SIP FW: 9.4(2) SR4-3
FreePBX IP: 172.17.50.99
Phone IP: 172.17.50.111 (DHCP)

I have a Cisco 7975G running the latest SIP Firmware version, and I have a xml file that is parsed ok and hands the settings to the phone, but for some reason the phone will not get past the “Registering…” phase

After several hours of playing with this I am completely out of ideas on how to fix this. There are some mentions on here form 5+ years ago about setting NAT to “Never” in freePBX somewhere, though non are clear where, and the only place I can find it is in the Asterisk SIP Settings (I am using pjSip) and toggling that does not make a difference.

I also tried using firmware 8.5(4)S with the exact same results.

Below are the logs from the phone and the XML config file.

Thanks in advance for the help.

Here are the pjSIP logs from asterisk: https://pastebin.freepbx.org/view/07d5a4e6

Here are the logs from the phone itself:

8399: WRN 21:15:30.090798 JVM: Startup Module Loader|cip.l10n.NetworkLocaleProperty:? - Unable to process LocaleProperty 'device.settings.config.localization.networklocale'
8400: WRN 21:15:30.175801 SECD: WARN:lookupCTL: ** no CTL, assume UCM NONSECURE
8401: ERR 21:15:30.524295 JVM: ..LOCALCONNECT FAILED
8402: NOT 21:15:30.581286 JVM: SIPCC-SIP_CTRL: sip_restart: In sip_restart
8403: NOT 21:15:30.592864 JVM: set_active_ccm: ccm=PRIMARY_CCM  port=0
8404: ERR 21:15:30.593596 JVM: SIP : sip_transport_setup_cc_conn : Admin has not configured a valid cucm for cucm index=SECONDARY_CCM=1.
8405: ERR 21:15:30.594367 JVM: SIP : sip_transport_setup_cc_conn : Admin has not configured a valid cucm for cucm index=TERTIARY_CCM=2.
8406: ERR 21:15:30.595060 JVM: SIP : sip_transport_setup_cc_conn : Admin has not configured a valid cucm for cucm index=SRST_CCM=3.
8407: ERR 21:15:30.595758 JVM: sip_regmgr_setup_cc_conns: NO VALID STANDBY CALL CONTROL AVAILABLE!
8408: NOT 21:15:30.596465 JVM: SIPCC-SIP_REG: sip_sm_init: Disabling mass reg state
8409: NOT 21:15:30.615824 JVM: SIPCC-SIP_REG: ccsip_register_all_lines: Disabling mass reg state
8410: NOT 21:15:30.616876 JVM: SIPCC-UI_API: 1/0, ui_set_sip_registration_state: 0
8411: NOT 21:15:30.619856 JVM: SIPCC-CC_API: 0/0, cc_int_feature: UI -> GSM: UPD_MEDIA_CAP       
8412: NOT 21:15:30.620963 JVM: SIPCC-SIP_CTRL: sip_restart: sip.taskInited is set to true 
8413: NOT 21:15:30.621763 JVM: SIPCC-CC_API: 0/0, cc_int_fail_fallback: SIP -> GSM: FAILOVER_FALLBACK   
8414: NOT 21:15:30.625160 JVM: SIPCC-DCSM: dcsm_process_event: DCSM 0   :(DCSM_READY:FEAT:UPD_MEDIA_CAP)
8415: NOT 21:15:30.627392 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_IDLE <- E_SIP_REG_REG_REQ
8416: NOT 21:15:30.629308 JVM: ccsip_messaging: sipSPIAddContactHeader: CFGID_DEVICE_NAME = SEP64D989C242E5
8417: NOT 21:15:30.632114 JVM: SIPCC-SIP_MSG_SEND: ccsip_dump_send_msg_info: <172.17.50.99:5060>:REGISTE: <sip:1001@172.17.50.99> :109 REGISTER::64d989c2-42e5000a-fd20cf24-af188fca@172.17.50.111
8418: NOT 21:15:30.832240 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter:  - cmname=172.17.50.99 cmIp=172.17.50.99 port=5060 isValid=true
8419: ERR 21:15:30.834325 JVM: Startup Module Loader|cip.sipcc.SipCcAdapter:  - addSrstToCAgentList: isSrstSecure=false
8420: NOT 21:15:30.836001 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0
8421: NOT 21:15:30.837928 JVM: Startup Module Loader|cip.sipcc.d:  - initializeLinePlane(): Mgmt Interface is in Service now..
8422: WRN 21:15:30.839600 JVM: Startup Module Loader|DisplayTask:? -  Line 1 property added with dn 1001
8423: ERR 21:15:30.841346 JVM: Startup Module Loader|cip.props.u:? - device.settings.energywise.keywords IO error.
8424: ERR 21:15:30.899385 JVM: tftpClient dialplan.xml /usr/cache/DP158232698 550001 1
8425: NOT 21:15:30.903249 JVM: Startup Module Loader|cip.cfg.ConfigManager:? - --->ConfigManager PropertyChanged: device.callagent.callcount
8426: NOT 21:15:30.904853 JVM: Startup Module Loader|cip.cfg.ConfigManager:? - <---ConfigManager PropertyChanged: device.callagent.callcount
8427: NOT 21:15:30.906494 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0
8428: NOT 21:15:30.907867 xxtpClient: xxtp request rcv'd from /usr/tmp/tftp, srcFile = dialplan.xml, dstFile = /usr/cache/DP158232698 max size = 550001 
8429: NOT 21:15:30.908967 ESP: server 1 =  
8430: NOT 21:15:30.922136 xxtpClient: auth server - tftpList[0] = ::ffff:172.17.50.99 
8431: NOT 21:15:30.922768 xxtpClient: look up server - 0 
8432: WRN 21:15:30.924796 SECD: WARN:lookupCTL: ** no CTL, assume TFTP NONSECURE
8433: NOT 21:15:30.927613 xxtpClient: secVal = 0xa 
8434: NOT 21:15:30.928332 xxtpClient: ::ffff:172.17.50.99 is a NONsecure server 
8435: NOT 21:15:30.928941 xxtpClient: temp retval = SRVR_NONSECURE, keep looking 
8436: NOT 21:15:30.929489 xxtpClient: retval = 10 
8437: NOT 21:15:30.930834 xxtpClient: Secure file requested 
8438: NOT 21:15:30.931487 xxtpClient: Non secure file approved  -- dialplan.xml  
8439: NOT 21:15:30.949471 HTTPCL: downdload will be limited to 537 KB
8440: ERR 21:15:30.955011 HTTPCL: connect() failed
8441: NOT 21:15:30.964677 SYSMSG: pid 29 (/sbin/httpcl) Normal Exit, status = 102
8442: INF 21:15:30.964717           runtime = 0.020 secs

8443: INF 21:15:30.964738          user cpu = 0.001228030 secs

8444: INF 21:15:30.964758        system cpu = 0.012194720 secs

8445: INF 21:15:30.964772    child user cpu = 0.000000000 secs

8446: INF 21:15:30.964787     child sys cpu = 0.000000000 secs

8447: INF 21:15:30.964808    sys interrupts = 0.000854890 secs for 22 interrupts

8448: INF 21:15:30.964832 total cpu = 0.013422750 secs ( 50% utilization )

8449: WRN 21:15:30.983327 xxtpClient: HTTP failed with code 102
8450: NOT 21:15:31.029141 TFTP: [28]:Requesting dialplan.xml from 172.17.50.99 with size limit of 550001
8451: NOT 21:15:31.032485 TFTP: [28]:Finished --> rcvd 33 bytes 
8452: NOT 21:15:31.122562 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_REGISTERING <- E_SIP_REG_TMR_RETRY
8453: NOT 21:15:31.172547 xxtpClient: request server6 0 ---> :: 
8454: NOT 21:15:31.173733 ESP: server 2 = :: 
8455: NOT 21:15:31.195827 xxtpClient: request server6 0 ---> :: 
8456: NOT 21:15:31.212366 xxtpClient: request server6 1 ---> :: 
8457: NOT 21:15:31.213556 ESP: server 3 = :: 
8458: NOT 21:15:31.221283 ESP: send ADMIN, logging = 1, shell = 0, ipconfig = 1 
8459: NOT 21:15:31.236792 xxtpClient: request server6 1 ---> :: 
8460: ERR 21:15:31.247187 JVM: ..LOCALCONNECT FAILED
8461: ERR 21:15:31.250871 JVM: ..LOCALCONNECT FAILED
8462: ERR 21:15:31.254280 JVM: ..LOCALCONNECT FAILED
8463: ERR 21:15:31.257753 JVM: ..LOCALCONNECT FAILED
8464: ERR 21:15:31.262039 JVM: ..LOCALCONNECT FAILED
8465: ERR 21:15:31.265428 JVM: ..LOCALCONNECT FAILED
8466: ERR 21:15:31.268844 JVM: ..LOCALCONNECT FAILED
8467: WRN 21:15:31.287606 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//CTLFile.tlv
8468: WRN 21:15:31.288627 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//ITLFile.tlv
8469: WRN 21:15:31.293952 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//CTLFile.tlv
8470: WRN 21:15:31.294965 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//ITLFile.tlv
8471: WRN 21:15:31.300194 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//CTLFile.tlv
8472: WRN 21:15:31.301209 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//ITLFile.tlv
8473: WRN 21:15:31.306795 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//CTLFile.tlv
8474: WRN 21:15:31.307865 SECD: WARN:getTLInfoFromFile: ** phone has no TL file /flash0/sec/ctl//ITLFile.tlv
8475: NOT 21:15:31.319010 xxtpClient: request server 0 ---> 172.17.50.99 
8476: NOT 21:15:31.319851 ESP: server 0 = 172.17.50.99 
8477: NOT 21:15:31.342964 xxtpClient: request server 1 --->  
8478: NOT 21:15:31.344121 ESP: server 1 =  
8479: NOT 21:15:31.366526 xxtpClient: request server 0 ---> 172.17.50.99 
8480: NOT 21:15:31.382980 xxtpClient: request server6 0 ---> :: 
8481: NOT 21:15:31.384230 ESP: server 2 = :: 
8482: NOT 21:15:31.415999 xxtpClient: request server 1 --->  
8483: NOT 21:15:31.435799 xxtpClient: request server6 1 ---> :: 
8484: NOT 21:15:31.436924 ESP: server 3 = :: 
8485: NOT 21:15:31.452302 CDP-D: cdpGetPortCfg SPANTOPC CFG:11
8486: NOT 21:15:31.545590 xxtpClient: request server6 0 ---> :: 
8487: NOT 21:15:31.566701 xxtpClient: request server6 1 ---> :: 
8488: ERR 21:15:31.572030 JVM: ..LOCALCONNECT FAILED
8489: ERR 21:15:31.575496 JVM: ..LOCALCONNECT FAILED
8490: ERR 21:15:31.578890 JVM: ..LOCALCONNECT FAILED
8491: ERR 21:15:31.583192 JVM: ..LOCALCONNECT FAILED
8492: ERR 21:15:31.586529 JVM: ..LOCALCONNECT FAILED
8493: ERR 21:15:31.589907 JVM: ..LOCALCONNECT FAILED
8494: ERR 21:15:31.593390 JVM: ..LOCALCONNECT FAILED
8495: NOT 21:15:32.122323 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_REGISTERING <- E_SIP_REG_TMR_RETRY
8496: NOT 21:15:34.126417 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_REGISTERING <- E_SIP_REG_TMR_RETRY
8497: NOT 21:15:38.122252 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_REGISTERING <- E_SIP_REG_TMR_RETRY
8498: NOT 21:15:42.122298 JVM: SIPCC-SIP_REG_STATE: 1/51, sip_reg_sm_process_event: SIP_REG_STATE_REGISTERING <- E_SIP_REG_TMR_RETRY

And here is my XML settings file:

<?xml version="1.0" encoding="UTF-8"?>
<device>
  <fullConfig>true</fullConfig>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>admin</sshUserId>
  <sshPassword>cisco</sshPassword>
  <devicePool>
                <dateTimeSetting>
                        <dateTemplate>M/D/Ya</dateTemplate>
                        <timeZone>Pacific Standard/Daylight Time</timeZone>
                        <ntps>
                                <ntp>
                                        <name>172.17.50.99</name>
                                        <ntpMode>Unicast</ntpMode>
                                </ntp>
                        </ntps>
                </dateTimeSetting>

     <callManagerGroup>
        <members>
           <member priority="0">
              <callManager>
                 <name>172.17.50.99</name>
                 <description>1800pbx</description>
                 <ports>
                    <ethernetPhonePort>2000</ethernetPhonePort>
                    <sipPort>5060</sipPort>
                    <securedSipPort>5061</securedSipPort>
                 </ports>
                 <processNodeName>172.17.50.99</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
           <connectionMonitorDuration>120</connectionMonitorDuration>
  </devicePool>

  <commonProfile>
     <phonePassword></phonePassword>
     <backgroundImageAccess>true</backgroundImageAccess>
     <callLogBlfEnabled>2</callLogBlfEnabled>
  </commonProfile>

  <loadInformation></loadInformation>

  <vendorConfig>
     <disableSpeaker>false</disableSpeaker>
     <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
     <pcPort>0</pcPort>
     <settingsAccess>1</settingsAccess>
     <garp>0</garp>
     <voiceVlanAccess>0</voiceVlanAccess>
     <videoCapability>1</videoCapability>
     <autoSelectLineEnable>0</autoSelectLineEnable>
     <webAccess>0</webAccess>
     <spanToPCPort>1</spanToPCPort>
     <loggingDisplay>1</loggingDisplay>
     <loadServer></loadServer>
     <daysDisplayNotActive></daysDisplayNotActive>
     <displayOnTime>07:00</displayOnTime>
     <displayOnDuration>17:00</displayOnDuration>
     <displayIdleTimeout>1:00</displayIdleTimeout>
  </vendorConfig>

  <userLocale>
       <name>English_United_States</name>
       <uid>1</uid>
       <langCode>en_US</langCode>
       <version>1.0.0.0-1</version>
       <winCharSet>iso-8859-1</winCharSet>
  </userLocale>
   <networkLocale>United_States</networkLocale>
   <networkLocaleInfo>
     <name>United_States</name>
     <version>1.0.0.0-1</version>
   </networkLocaleInfo>

   <deviceSecurityMode>1</deviceSecurityMode>
   <authenticationURL>http://172.17.50.99/cisco/services/authentication.php</authenticationURL>
   <directoryURL>http://172.17.50.99/xmlservices/PhoneDirectory.php</directoryURL>
   <idleURL>http://172.17.50.99/xmlservices/index.php</idleURL>
   <informationURL>http://172.17.50.99/xmlservices/help.php</informationURL>
   <messagesURL></messagesURL>
   <proxyServerURL></proxyServerURL>
   <servicesURL>http://172.17.50.99/xmlservices/index.php</servicesURL>
   <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
   <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
   <dscpForCm2Dvce>96</dscpForCm2Dvce>
   <transportLayerProtocol>2</transportLayerProtocol>
   <dndCallAlert>5</dndCallAlert>
   <capfAuthMode>0</capfAuthMode>
   <capfList>
      <capf>
         <phonePort>3804</phonePort>
      </capf>
   </capfList>

   <certHash></certHash>
   <encrConfig>false</encrConfig>

   <sipProfile>
     <sipProxies>
        <backupProxy></backupProxy>
        <backupProxyPort>5060</backupProxyPort>
        <emergencyProxy></emergencyProxy>
        <emergencyProxyPort>5060</emergencyProxyPort>
        <outboundProxy></outboundProxy>
        <outboundProxyPort>5060</outboundProxyPort>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>  
   <sipCallFeatures>
       <cnfJoinEnabled>true</cnfJoinEnabled>
       <callForwardURI>x--serviceuri-cfwdall</callForwardURI>
       <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
       <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
       <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
       <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
       <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
       <rfc2543Hold>false</rfc2543Hold>
       <callHoldRingback>2</callHoldRingback>
       <localCfwdEnable>true</localCfwdEnable>
       <semiAttendedTransfer>true</semiAttendedTransfer>
       <anonymousCallBlock>2</anonymousCallBlock>
       <callerIdBlocking>2</callerIdBlocking>
       <dndControl>0</dndControl>
       <remoteCcEnable>true</remoteCcEnable>
   </sipCallFeatures>

   <sipStack>
        <sipInviteRetx>6</sipInviteRetx>
        <sipRetx>10</sipRetx>
        <timerInviteExpires>180</timerInviteExpires>
        <timerRegisterExpires>3600</timerRegisterExpires>
        <timerRegisterDelta>5</timerRegisterDelta>
        <timerKeepAliveExpires>120</timerKeepAliveExpires>
        <timerSubscribeExpires>120</timerSubscribeExpires>
        <timerSubscribeDelta>5</timerSubscribeDelta>
        <timerT1>500</timerT1>
        <timerT2>4000</timerT2>
        <maxRedirects>70</maxRedirects>
        <remotePartyID>false</remotePartyID>
        <userInfo>None</userInfo>
   </sipStack>

     <autoAnswerTimer>1</autoAnswerTimer>
     <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
     <autoAnswerOverride>true</autoAnswerOverride>
     <transferOnhookEnabled>false</transferOnhookEnabled>
     <enableVad>false</enableVad>
     <preferredCodec>none</preferredCodec>
     <dtmfAvtPayload>101</dtmfAvtPayload>
     <dtmfDbLevel>3</dtmfDbLevel>
     <dtmfOutofBand>avt</dtmfOutofBand>
     <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
     <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
     <kpml>3</kpml>
     <natEnabled>false</natEnabled>
     <natAddress></natAddress>
     <stutterMsgWaiting>0</stutterMsgWaiting>
     <callStats>false</callStats>
     <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
     <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
     <startMediaPort>16384</startMediaPort>
     <stopMediaPort>32766</stopMediaPort>
     <voipControlPort>5060</voipControlPort>
     <dscpForAudio>184</dscpForAudio>
     <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
     <dialTemplate>dialplan.xml</dialTemplate>

         <phoneLabel>Office</phoneLabel>
     <sipLines>
        <line button="1">
           <featureID>9</featureID>
           <featureLabel>1001</featureLabel>
           <name>1001</name>
           <displayName>1001</displayName>
           <contact>1001</contact>
           <proxy>USECALLMANAGER</proxy>
           <port>5060</port>
           <autoAnswer>
              <autoAnswerEnabled>2</autoAnswerEnabled>
           </autoAnswer>
           <callWaiting>3</callWaiting>
           <authName>1001</authName>
           <authPassword>{8 character password}</authPassword>
           <sharedLine>false</sharedLine>
           <messageWaitingLampPolicy>1</messageWaitingLampPolicy>
           <messagesNumber>*97</messagesNumber>
           <ringSettingIdle>4</ringSettingIdle>
           <ringSettingActive>5</ringSettingActive>
           <forwardCallInfoDisplay>
              <callerName>true</callerName>
              <callerNumber>false</callerNumber>
              <redirectedNumber>false</redirectedNumber>
              <dialedNumber>true</dialedNumber>
           </forwardCallInfoDisplay>
        </line>
     </sipLines>
  </sipProfile>
</device>

Posts: 7

Participants: 2

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@chris_unit wrote:

Hi there

Just setting up my FreePBX 14

Zulu Desktop works between two laptops very well but i think Zulu connects using different ports/protocols.

When i try to call from my Android using Grandstream Wave using pjsip i can hear ringing but no audio when the call connects.

I dont think this is NAT because all devices are on the same lan

Posts: 1

Participants: 1

Read full topic

@mmguglielmi wrote:

Hi everybody! Thnxs for reading in advance
I’m driving me crazy, I can’t find a solution for my problem. I’ve struggled long hours, so I decided to ask for help

At first, I’ll tell you THE problem, all working ok, but incoming calls end because of nat (I think) after 30 seconds.
Asterisk has 2 network boards, static public ip, extensions connected from outside the site. One of the boards is connected to a sip trunk, I’ve NAT enabled. Audio for both sides
In a SIP debug, I see “Retransmitting #10 (NAT) to 111.111.3.10:5060:” in INVITEs, but I think that is wrong the contact info (I see the public IP, instead the eth1’s IP), so I think it is the problem, and I don’t know how to solve it

Hope somebody can help me

Thnx again

I changed IPs, just for security

Asterisk 11.25.3

2 network boards (eth0 network/internet, eth1 sip trunk)

eth0 Link encap:Ethernet HWaddr 78:2B:CB:AE:0E:16
inet addr:192.168.10.223 Bcast:192.168.10.255 Mask:255.255.255.0
inet6 addr: xxxx::7a2b:cbff:feae:e16/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:98692 errors:0 dropped:0 overruns:0 frame:0
TX packets:85077 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:16984481 (16.1 MiB) TX bytes:73086281 (69.7 MiB)
Interrupt:20 Memory:e1c00000-e1c20000

eth1 Link encap:Ethernet HWaddr 7C:8B:CA:00:3D:7C
inet addr:111.111.64.149 Bcast:111.111.64.151 Mask:255.255.255.252
inet6 addr: xxxx::7e8b:caff:fe00:3d7c/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:29770 errors:0 dropped:0 overruns:0 frame:0
TX packets:27269 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:6193913 (5.9 MiB) TX bytes:6111981 (5.8 MiB)

Trunk name: Telec
host=111.111.3.10
type=peer
context=from-trunk
fromdomain=111.111.64.149
disallow=all
allow=alaw

Public IP 222.222.234.123

route -n

Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
111.111.3.10 111.111.64.150 255.255.255.255 UGH 0 0 0 eth1
111.111.64.148 0.0.0.0 255.255.255.252 U 0 0 0 eth1
192.168.10.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 eth1
169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0 0 eth0
0.0.0.0 192.168.10.1 0.0.0.0 UG 0 0 0 eth0

---

sip.conf

nat=yes
ALLOW_SIP_ANON=no
externip=222.222.234.123
localnet=192.168.10.0/24
localnet=111.111.64.148/30

---

SIP debug, an external call from 1166667777 to 1122223333, ext 4466

[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – Called SIP/4466
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – Connected line update to SIP/Telec-0000000e prevented.
[2020-06-04 20:19:39] VERBOSE[3787][C-0000000e] app_dial.c: – SIP/4466-0000000f is ringing
[2020-06-04 20:19:42] VERBOSE[3787][C-0000000e] app_dial.c: – Connected line update to SIP/Telec-0000000e prevented.
[2020-06-04 20:19:42] VERBOSE[3787][C-0000000e] app_dial.c: – SIP/4466-0000000f answered SIP/Telec-0000000e
[2020-06-04 20:19:46] VERBOSE[2409] chan_sip.c: Retransmitting #5 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:19:50] VERBOSE[2409] chan_sip.c: Retransmitting #6 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:19:54] VERBOSE[2409] chan_sip.c: Retransmitting #7 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:19:58] VERBOSE[2409] chan_sip.c: Retransmitting #8 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:20:00] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.252:5060 —>

<------------->
[2020-06-04 20:20:02] VERBOSE[2409] chan_sip.c: Retransmitting #9 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:20:03] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘0gQAAC8WAAACBAAALxYAAJa26xmPlKr8wixeofhzyRdMoaLor7v3oik715/n3IFF@111.111.3.10’ Method: OPTIONS
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:111.111.3.10:5060 —>
OPTIONS sip:metaswitch@111.111.64.149:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+2183e4ea3ff7dd836b11c1e3bf15c7731+sip+2+b00889c8
From: sip:metaswitch@111.111.3.10:5060;tag=111.111.3.10+2+6eb16302+c68d2e13
Content-Length: 0
Supported: resource-priority, siprec, 100rel
To: sip:metaswitch@111.111.64.149
Contact: sip:941235c91a33e3b43cf7d85de76db36e@111.111.3.10
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name
Max-Forwards: 69
Call-ID: 0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10
CSeq: 699873511 OPTIONS
Organization: Metaswitch Networks
Accept: application/sdp, application/dtmf-relay

<------------->
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: — (13 headers 0 lines) —
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Sending to 111.111.3.10:5060 (NAT)
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Looking for metaswitch in from-sip-external (domain 111.111.64.149)
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 111.111.3.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+2183e4ea3ff7dd836b11c1e3bf15c7731+sip+2+b00889c8;received=111.111.3.10;rport=5060
From: sip:metaswitch@111.111.3.10:5060;tag=111.111.3.10+2+6eb16302+c68d2e13
To: sip:metaswitch@111.111.64.149;tag=as371684c3
Call-ID: 0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10
CSeq: 699873511 OPTIONS
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:222.222.234.123:5060
Accept: application/sdp
Content-Length: 0

<------------>
[2020-06-04 20:20:05] VERBOSE[2409] chan_sip.c: Scheduling destruction of SIP dialog ‘0gQAAC8WAAACBAAALxYAAAF+Gn9XKilc3HNYnZtDoflCIM8wf/st6To/NgxdE2mM@111.111.3.10’ in 32000 ms (Method: OPTIONS)
[2020-06-04 20:20:06] VERBOSE[2409] chan_sip.c: Retransmitting #10 (NAT) to 111.111.3.10:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.3.10:5060;branch=z9hG4bK+86e463137aecf9f409ba8a15f76dc3591+sip+6+aff49b2b;received=111.111.3.10;rport=5060
From: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
To: sip:1122223333@111.111.64.149;tag=as2e963a0b
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 56524576 INVITE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1122223333@222.222.234.123:5060
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 190755527 190755527 IN IP4 222.222.234.123
s=Asterisk PBX 11.25.3
c=IN IP4 222.222.234.123
t=0 0
m=audio 10940 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

[2020-06-04 20:20:07] WARNING[2409] chan_sip.c: Retransmission timeout reached on transmission 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10 for seqno 56524576 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[2020-06-04 20:20:07] WARNING[2409] chan_sip.c: Hanging up call 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [h@macro-dial-one:1] Macro(“SIP/Telec-0000000e”, “hangupcall,”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:1] ExecIf(“SIP/Telec-0000000e”, “0?Set(CDR(recordingfile)=.wav)”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:2] GotoIf(“SIP/Telec-0000000e”, “1?theend”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Goto (macro-hangupcall,s,4)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/Telec-0000000e”, “”) in new stack
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘hangupcall’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/Telec-0000000e’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Scheduling destruction of SIP dialog ‘2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060’ in 6400 ms (Method: INVITE)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: Parsing sip:4466@192.168.10.36:60798 for address/port to send to
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: set destination to 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.36:60798:
BYE sip:4466@192.168.10.36:60798 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK2a54c60e;rport
Max-Forwards: 70
From: “1166667777” sip:1166667777@192.168.10.223;tag=as6604fad6
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=aa39c34a
Call-ID: 2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060
CSeq: 103 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-dial-one, s, 45) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘dial-one’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] app_macro.c: == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/Telec-0000000e’ in macro ‘exten-vm’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] pbx.c: == Spawn extension (ext-local, 4466, 2) exited non-zero on ‘SIP/Telec-0000000e’
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Scheduling destruction of SIP dialog ‘0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10’ in 32000 ms (Method: INVITE)
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: Parsing sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10 for address/port to send to
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: set_destination: set destination to 111.111.3.10:5060
[2020-06-04 20:20:07] VERBOSE[3787][C-0000000e] chan_sip.c: Reliably Transmitting (NAT) to 111.111.3.10:5060:
BYE sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10 SIP/2.0
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK42f5707d;rport
Max-Forwards: 70
From: sip:1122223333@111.111.64.149;tag=as2e963a0b
To: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 102 BYE
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

---

[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK2a54c60e;rport=5060
Contact: sip:4466@192.168.10.36:60798
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=aa39c34a
From: “1166667777” sip:1166667777@192.168.10.223;tag=as6604fad6
Call-ID: 2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060
CSeq: 103 BYE
User-Agent: Z 3.15.40006 rv2.8.20
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (9 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘2df5b9a014dafb8862ec863e3cdafbe2@192.168.10.223:5060’ Method: INVITE
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:111.111.3.10:5060 —>
SIP/2.0 200 OK
Call-ID: 0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10
CSeq: 102 BYE
From: sip:1122223333@111.111.64.149;tag=as2e963a0b
To: sip:1166667777@111.111.3.10:5060;tag=111.111.3.10+6+d582edab+d0c78e7e
Via: SIP/2.0/UDP 222.222.234.123:5060;received=111.111.64.149;rport=5060;branch=z9hG4bK42f5707d
Content-Length: 0
Supported: resource-priority, siprec, 100rel
Contact: sip:4876c0435de2db36c27a315bebe64c62@111.111.3.10
Server: DC-SIP/2.0
Organization: Metaswitch Networks
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (12 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409][C-0000000e] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘0gQAAC8WAAACBAAALxYAABKW5K4dBEPAqJ9GNe/2xO8AN67To034BFUOE0iRoDID@111.111.3.10’ Method: INVITE
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
PUBLISH sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—ba8a6e7aed23642f
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=0b20e91e
Call-ID: y-qUP1725REoFNjQnRZg0Q…
CSeq: 1 PUBLISH
Expires: 600
Content-Type: application/pidf+xml
User-Agent: Z 3.15.40006 rv2.8.20
Event: presence
Allow-Events: presence, kpml, talk
Content-Length: 262

<?xml version="1.0" encoding="UTF-8"?>

open Online

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (14 headers 3 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—ba8a6e7aed23642f;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=0b20e91e
To: sip:4466@192.168.10.223;transport=UDP;tag=as5edf2150
Call-ID: y-qUP1725REoFNjQnRZg0Q…
CSeq: 1 PUBLISH
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘y-qUP1725REoFNjQnRZg0Q…’ Method: PUBLISH
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SUBSCRIBE sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—7acf366feff11173
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
User-Agent: Z 3.15.40006 rv2.8.20
Event: presence.winfo
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Creating new subscription
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: list_route: hop: sip:4466@192.168.10.36:60798;transport=UDP
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Found peer ‘4466’ for ‘4466’ from 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—7acf366feff11173;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
To: sip:4466@192.168.10.223;transport=UDP;tag=as7dc61397
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 1 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7516b3e8”
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Scheduling destruction of SIP dialog ‘QqCMBDM_dVMSM5fZ5HyN2A…’ in 6400 ms (Method: SUBSCRIBE)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SUBSCRIBE sip:4466@192.168.10.223;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—d7a542af6bfdfbb6
Max-Forwards: 70
Contact: sip:4466@192.168.10.36:60798;transport=UDP
To: sip:4466@192.168.10.223;transport=UDP
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
User-Agent: Z 3.15.40006 rv2.8.20
Authorization: Digest username=“4466”,realm=“asterisk”,nonce=“7516b3e8”,uri="sip:4466@192.168.10.223;transport=UDP",response=“9aac98c51da5114be076284f7b0b3393”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: — (15 headers 0 lines) —
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Creating new subscription
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Sending to 192.168.10.36:60798 (NAT)
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Found peer ‘4466’ for ‘4466’ from 192.168.10.36:60798
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c:
<— Transmitting (NAT) to 192.168.10.36:60798 —>
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.10.36:60798;branch=z9hG4bK-524287-1—d7a542af6bfdfbb6;received=192.168.10.36;rport=60798
From: sip:4466@192.168.10.223;transport=UDP;tag=4e63d737
To: sip:4466@192.168.10.223;transport=UDP;tag=as7dc61397
Call-ID: QqCMBDM_dVMSM5fZ5HyN2A…
CSeq: 2 SUBSCRIBE
Server: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2020-06-04 20:20:07] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘QqCMBDM_dVMSM5fZ5HyN2A…’ Method: SUBSCRIBE
[2020-06-04 20:20:08] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.132:5060 —>

<------------->
[2020-06-04 20:20:08] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘D4BFEBF0-3@111.111.3.10:5060’ Method: OPTIONS
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 24.232.134.32:55974:
OPTIONS sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK7f1ffec9;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@222.222.234.123;tag=as641529a1
To: sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP
Contact: sip:Unknown@222.222.234.123:5060
Call-ID: 229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:24.232.134.32:55974 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.222.234.123:5060;branch=z9hG4bK7f1ffec9;rport=5060
Contact: sip:24.232.134.32:55974
To: sip:4444@24.232.134.32:55974;rinstance=272c0387725cd0b8;transport=UDP;tag=dc3f5348
From: "Unknown"sip:Unknown@222.222.234.123;tag=as641529a1
Call-ID: 229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘229901085ddfcdf923b1799b3de772e8@222.222.234.123:5060’ Method: OPTIONS
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.36:60798:
OPTIONS sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK04d8555b;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.10.223;tag=as1f1562f8
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP
Contact: sip:Unknown@192.168.10.223:5060
Call-ID: 4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.36:60798 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK04d8555b;rport=5060
Contact: sip:192.168.10.36:60798
To: sip:4466@192.168.10.36:60798;rinstance=b84ac64a2db3b64f;transport=UDP;tag=f45e8623
From: “Unknown” sip:Unknown@192.168.10.223;tag=as1f1562f8
Call-ID: 4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: — (14 headers 0 lines) —
[2020-06-04 20:20:12] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘4b8f5a6e098f932743c9dcc530ac8957@192.168.10.223:5060’ Method: OPTIONS
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: Reliably Transmitting (NAT) to 192.168.10.252:5060:
OPTIONS sip:900@192.168.10.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK7c81c2d7;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@192.168.10.223;tag=as2a1cc252
To: sip:900@192.168.10.252:5060
Contact: sip:Unknown@192.168.10.223:5060
Call-ID: 78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-AsteriskNOW-12.0.76.6(11.25.3)
Date: Thu, 04 Jun 2020 23:20:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c:
<— SIP read from UDP:192.168.10.252:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.223:5060;branch=z9hG4bK7c81c2d7;rport=5060
From: “Unknown” sip:Unknown@192.168.10.223;tag=as2a1cc252
To: sip:900@192.168.10.252:5060;tag=3991286321
Call-ID: 78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.130
Content-Length: 0

<------------->
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: — (8 headers 0 lines) —
[2020-06-04 20:20:13] VERBOSE[2409] chan_sip.c: Really destroying SIP dialog ‘78aee7092263ff811bf07c1b1c2db4a1@192.168.10.223:5060’ Method: OPTIONS

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@chris_unit wrote:

https://pastebin.com/VunEBqPM

i’m getting dropped calls after about 30 seconds between an android using GS Wave and OSX using Zulu.

ringing and voice works but only for about 30 seconds then the call is dropped

log in pastebin

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@faze wrote:

Hi,
I’ve got a few pjsip extensions registered using freepbx.
Here’s the (partial) output from pjsip show endpoints:

 Endpoint:  100/100                                              Not in use    0 of inf
     InAuth:  100-auth/100
        Aor:  100                                                1
      Contact:  100/sip:100@192.168.1.12:12410;transport=T 127383ff5c Avail        17.425

 Endpoint:  101/101                                              Not in use    0 of inf
     InAuth:  101-auth/101
        Aor:  101                                                1
      Contact:  101/sip:101@192.168.1.36:37249;transport=T c01a62a4d0 Avail         3.575

For some reason, internal calls do not work.

I have so far:

• checked that both extens use the context from-internal
• changed my outbound route dialplan from X. to XXXX., because previously these internal calls would be routed out via a trunk
• tried calling both 101 and (with prefix)*101

I can see in asterisk that the following dialplan function is triggered:

-- Executing [100@from-internal:5] Playback("PJSIP/101-0000000f", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack

I’d appreciate any ideas how to fix this.

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