@ernied wrote:

Every day for the past three or four days, we’ve had to restart Asterisk manually from the command line (using /etc/init.d/asterisk) , because it had stopped taking calls. This has also been a problem in the past, but this is the last straw.

Restarting Asterisk restores normal service, but this is extremely disruptive, as calls get dropped, or new calls get rejected until someone takes the time to fix this.

The only logs I’ve seen that occur around this time are certain PJSIP users repeatedly attempting to log in and failing. Like so:

[2018-07-05 09:52:26] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1050” sip:1050@192.168.0.7’ failed for ‘192.168.0.117:5060’ (callid: 0_695042223@192.168.0.117) - Failed to authenticate
[2018-07-05 09:52:26] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1050” sip:1050@192.168.0.7’ failed for ‘192.168.0.117:5060’ (callid: 0_695042223@192.168.0.117) - Failed to authenticate
[2018-07-05 09:52:26] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1050” sip:1050@192.168.0.7’ failed for ‘192.168.0.117:5060’ (callid: 0_695042223@192.168.0.117) - Failed to authenticate
[2018-07-05 09:52:26] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1050” sip:1050@192.168.0.7’ failed for ‘192.168.0.117:5060’ (callid: 0_695042223@192.168.0.117) - Failed to authenticate
[2018-07-05 09:52:29] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1049” sip:1049@192.168.0.7’ failed for ‘192.168.0.77:5060’ (callid: 0_839451309@192.168.0.77) - Failed to authenticate
[2018-07-05 09:52:29] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1049” sip:1049@192.168.0.7’ failed for ‘192.168.0.77:5060’ (callid: 0_839451309@192.168.0.77) - Failed to authenticate
[2018-07-05 09:52:29] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1049” sip:1049@192.168.0.7’ failed for ‘192.168.0.77:5060’ (callid: 0_839451309@192.168.0.77) - Failed to authenticate
[2018-07-05 09:52:29] NOTICE[5173][C-00000000]: func_audiohookinherit.c:64 func_inheritance_write: AUDIOHOOK_INHERIT is deprecated and now does nothing.
[2018-07-05 09:52:29] NOTICE[4984]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“1049” sip:1049@192.168.0.7’ failed for ‘192.168.0.77:5060’ (callid: 0_839451309@192.168.0.77) - Failed to authenticate

On a possibly related note, when starting Asterisk, I also see several errors about certain Asterisk modules failing to load. Like this:

[2018-07-05 09:52:25] ERROR[4812]: cdr_syslog.c:145 load_config: Unable to load cdr_syslog.conf. Not logging custom CSV CDRs to syslog.

[2018-07-05 09:52:25] ERROR[4812]: app_amd.c:445 load_config: Configuration file amd.conf missing.
[2018-07-05 09:52:25] ERROR[4812]: chan_unistim.c:6757 reload_config: Unable to load config unistim.conf

This should be routine, but in the past I’ve resolved some Asterisk stability issues on other Asterisk servers by setting most modules to not load in /etc/asterisk/modules.conf. Since our FreePBX server is close to stock, I expect this is something that needs to be fixed for the benefit of other FreePBX users.

Other things to note: We currently have 32 hotdesk users and 21 SIP phones on our network. I expect this isn’t an unreasonable strain on a single FreePBX server.

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@caoquocai wrote:

Dear all,

I am trying to sync the caller name from other database to VoIP server.

I already success to update the name into asterisk database (tables: users, devices, sip).

the problem is, the calling name doesn’t display like the callerID that I synced. I already tried to reload the VoIP server (fwconsole reload).

when I tried “database show ampuser XXXX” the “/AMPUSER/XXXX/cidname” was not changed to the new callerID.

Thanks

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@normic wrote:

Hi,
we’ve bought and activated the Fax Pro module a few days ago. We also managed to send and receive faxes with it.

But I didn’t find a way to achieve the following:
We have one main faxnumber. The receiving part will be collected by the info@ email address.
But we like to allow some users to use the main faxnumber to send faxes. Preferably by including the fax in their own UCP.

Of course we can give them the credentials and have them log in as the fax user, but this is unconvenient for the users…

Would this be possible and how?

Thanks in advance.
BR
normic

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@gcataldo wrote:

Hi,
I will tell you about my installation procedure:

• Install FreePBX distro successful.
• I ran dahdi_hardware and all my voip hardware appears (OpenVox E1 card and OpenVox A800E card). I only have problems with A800E.
• Once I activated opvxa1200 kernel module on freePBX WEB UI. My OpenVox A800E card was not recognize by dahdi_scan
• I compiled and install OpenVox driver from OpenVox website
• Once I ran modprobe opvxa1200, my system halt inmediately

Please could you help me please?

Sincerely,
GC

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@mjb2000 wrote:

My ISP here in Mexico, Telmex, provides a Fiber Optic Optical Network Terminal (ONT) - Essentially a piece of Consumer Premises Equipment (CPE) that functions as a modem, router, switch, WiFi access point and crucially - as an ATA, connecting to the ISP via SIP and offering a phone line from the telephone company.

I was able to obtain the SIP credentials by connecting a serial console to my modem. I have been able to use these credentials successfully with PhonerLite, Zoiper and MicroSIP (for all of these software packages I can register and make phone calls.

TELMEX seems to use SRV records which I couldn’t get working with MicroSIP, so I substituted the proxy domain name with the IP address that PhonerLite was using and that meant I was able to get MicroSIP to register correctly.

Asterisk info shows “Request sent” for registration, but is never registers.

@Stewart1 was trying to help me on this thread, but suggesting creating a new thread specifically for my problem. One this he mentioned was “copying a register to a file and using sipsak”… This is above my level of understanding and I don’t know how I would go about this. Does anyone else know what might be going on here? How can I take my working MicroSIP config and use it successfully in FreePBX?

My current config is:

Outbound
Trunk Name: Telmex
PEER Details: [BLANK]
Inbound
USER Context: +52xxxxxxxxxx
USER Details:

secret=PASSWORD
username=+520000000000
user=+520000000000
fromuser=+520000000000
realm=ims.telmex.com
domain=ims.telmex.com
authdomain=ims.telmex.com
fromdomain=ims.telmex.com
outboundproxy=189.247.242.147
host=189.247.242.147
fullcontact=+520000000000@ims.telmex.com
authname=+520000000000@ims.telmex.com
type=peer
insecure=very
context=from-sip-external

Register string:
+520000000000@ims.telmex.com:PASSWORD:+520000000000@189.247.242.147

Asterisk SIP log…

*.*.*.* = my external IP address, not the IP of the FreePBX box (which is 10.0.0.32)

[2018-07-09 12:35:41] WARNING[1019] chan_sip.c: Section 'Telmex' lacks type
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: Really destroying SIP dialog '75db2cf62a7ed81e14c45f5e28956512@10.200.0.32' Method: REGISTER
[2018-07-09 12:35:41] NOTICE[1019] chan_sip.c: -- Re-registration for +520000000000@189.247.242.147
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: REGISTER 11 headers, 0 lines
[2018-07-09 12:35:41] VERBOSE[1019] chan_sip.c: Reliably Transmitting (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0


---
[2018-07-09 12:35:42] VERBOSE[1019] chan_sip.c: Retransmitting #1 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0


---
[2018-07-09 12:35:43] VERBOSE[1019] chan_sip.c: Retransmitting #2 (NAT) to 189.247.242.147:5060:
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP *.*.*.*:5160;branch=z9hG4bK4af132fc;rport
Max-Forwards: 70
From: <sip:+520000000000@ims.telmex.com>;tag=as3d7f6af0
To: <sip:+520000000000@ims.telmex.com>
Call-ID: 1cbc86d41b0e907b31bf29842558a31e@10.200.0.32
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.6(14.7.6)
Expires: 120
Contact: <sip:s@*.*.*.*:5160>
Content-Length: 0

PhonerLite config (working)

Proxy/Registrar: voipnvcompigl.telmex.net
Domain/Realm: ims.telmex.com
Username: +52xxxxxxxxxx
Password: ############
Authentication name: +52xxxxxxxxxx@ims.telmex.com

MicroSIP config (working)

SIP Server: ims.telmex.com
SIP Proxy: 189.247.242.147
Username: +52xxxxxxxxxx
Domain: ims.telmex.com
Login: +52xxxxxxxxxx@ims.telmex.com
Password: ############

PhonerLite log

From here you can see why I ended up using the proxy IP address

I am not sure what the various forbidden and timeout messages are, but PhonerLite does seem to work correctly (I can make and receive calls).

-------------------------------------------
09:52:17,558: R: DNS lookup for 'voipnvcompigl.telmex.net'
start resolving SRV (UDP)...
-------------------------------------------
09:52:17,561: R: DNS lookup for 'slbcompigl.voip.telmex.net'
189.247.242.147:5060 (TTL=1699)
-------------------------------------------
09:52:17,561: R: open UDP port (SIP): 5060

-------------------------------------------
09:52:17,562: R: open TCP port (TLS listen): 5061

-------------------------------------------
09:52:17,562: R: open TCP port (TCP listen): 5060

09:52:17,611: Listen Confirm: 0E 00 08 00 05 81 9E 02 01 00 00 00 00 00 
09:52:17,611: Listen Confirm
-------------------------------------------
09:52:17,563: R: open UDP port (mDNS): 5353

09:52:17,621: Facility Confirm: 1A 00 08 00 80 81 A0 02 01 00 00 00 00 00 03 00 09 00 00 06 00 00 3D 01 00 00 
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621: Facility Request: 16 00 08 00 80 80 A1 02 01 00 00 00 03 00 07 01 00 04 3D 01 00 00 
09:52:17,621: Facility Request (Listen To Supplementary Services)
09:52:17,621:  Get Supported Services: success
-------------------------------------------
09:52:17,563: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0


-------------------------------------------
09:52:17,563: T: mDNS refresh: sip:+52xxxxxxxxxx@ims.telmex.com = 169.254.106.102:5060, ttl=900
SIPPER for PhonerLite
09:52:17,621: Facility Confirm: 16 00 08 00 80 81 A1 02 01 00 00 00 00 00 03 00 05 01 00 02 00 00 
09:52:17,621: Facility Confirm (Supplementary Services)
09:52:17,621:  Listen: success
-------------------------------------------
09:52:17,621: T: 189.247.242.147:5060 (UDP)
SUBSCRIBE sip:+52xxxxxxxxxx@ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Max-Forwards: 70
User-Agent: SIPPER for PhonerLite
Expires: 1800
Event: message-summary
Accept: application/simple-message-summary
Content-Length: 0


-------------------------------------------
09:52:17,705: R: 189.247.242.147:5060 (UDP)
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bcbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=3576163364
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=aprqngfrt-cpkluc30080e4
Call-ID: 80A6192D-2C81-E811-87B9-BB0BE53ECAEB@10.0.0.50
CSeq: 327 SUBSCRIBE


-------------------------------------------
09:52:17,847: R: 189.247.242.147:5060 (UDP)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bbbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=dqdq0bbq
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 326 REGISTER
WWW-Authenticate: Digest realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==",algorithm=MD5
Content-Length: 0


-------------------------------------------
09:52:17,848: T: 189.247.242.147:5060 (UDP)
REGISTER sip:ims.telmex.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.50:5060;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Authorization: Digest username="+52xxxxxxxxxx@ims.telmex.com", realm="ims.telmex.com", nonce="PBk3fi0dSOVla8Cwre8YRw==", uri="sip:ims.telmex.com", response="bd636de3caf80c6c64ba2910c30555f9", algorithm=MD5
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
Allow-Events: org.3gpp.nwinitdereg
User-Agent: SIPPER for PhonerLite
Supported: replaces, from-change, gruu
Expires: 900
Content-Length: 0


-------------------------------------------
09:52:18,100: R: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK80a6192d2c81e81187bdbb0be53ecaeb;rport=60580
From: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=1934202406
To: <sip:+52xxxxxxxxxx@ims.telmex.com>;tag=k2dqaboc
Call-ID: 80A6192D-2C81-E811-87B8-BB0BE53ECAEB@10.0.0.50
CSeq: 328 REGISTER
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com;user=phone>
P-Associated-URI: <sip:+52xxxxxxxxxx@ims.telmex.com>
Accept-Resource-Priority: wps.4
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>;expires=30;q=1;+sip.instance="<urn:uuid:00D1773F-C27F-E811-97E2-EED72484AD05>"
Content-Length: 0


-------------------------------------------
09:52:18,123: T: 189.247.242.147:5060 (UDP)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.247.242.147:5060;branch=z9hG4bKrc0icv204g1ge1u3epu0.1
From: <sip:+52xxxxxxxxxx@ims.telmex.com:5060>;tag=2gdrr79g-CC-20
To: <sip:+52xxxxxxxxxx@ims.telmex.com:60580>;tag=003db22d2c81e81187bdbb0be53ecaeb
Call-ID: 8g323bee3s8749i8a4r9r9iaa2a34s93@19500.0.ATS.ats01.ims.telmex.com.20
CSeq: 1 NOTIFY
Contact: <sip:+52xxxxxxxxxx@10.0.0.50:5060>
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Server: SIPPER for PhonerLite
Content-Length: 0


-------------------------------------------
09:52:18,293: R: 189.247.242.147:5060 (UDP)
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.0.0.50:5060;received=*.*.*.*;branch=z9hG4bK.bbkAMUUuC;rport=60580
From: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=mTMk6w0eA
To: <sip:52xxxxxxxxxx@voipnvcompigl.telmex.net>;tag=uhf99zaa
CSeq: 210 REGISTER
Call-ID: grNrFpLzKe
Warning: 399 P.5.127.ims.telmex.com "SS170001F133L3261S0E0[00001] Hllm query failed"
Content-Length: 0

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@rnrstar wrote:

We have an HA cluster of two servers. We are seeing the swap memory growing and never gets lower. From the time we turned it on about a week ago, it has worked its way up to 33% swap memory used. This is a new configuration and as far as we can tell, everything else is working fine. We are just concerned that this may be exposing an underlying problem.

FreePBX 13.0.195.4
High Availability Services 13.0.11

Any suggestions?

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@Christopheric1 wrote:

I have some phones that are Cisco 6945’s and I have also purchased the endpoint manager for the Cisco’s. I had to use the 6921 template and add things via the “baseline edit” function of endpoint manager. That works great and the buttons show up for the 3 other lines. The issue I am having is that BLF is not working. I also get issues when setting up those soft keys as pickup, park, etc. It gives me the error that “Unable to create a call. The max number of calls on this line have been reached” or something to that effect.

I am not sure where to go from here or if there is a patch for freepbx that I can add to make SIP work on the Cisco’s? I don’t want to go the road of SCCP as that in my opinion is going backwards.

Any help would be great!

Thanks,

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@hansentech wrote:

We recently upgraded some FreePBX servers from version 12 to 14. Some remote Grandstream phones using an OpenVPN connection suddenly stopped working by not passing along audio to the remote phones in both intercom and external calls. The NAT settings are correct as all other phones work just fine with both internal and external calls. Interestingly enough, the same phones and configuration works with FreePBX 12, just not 14. Also, using an OpenVPN connection on a PC with the X-Lite softphone by CounterPath also works fine with FreePBX 14. The OpenVPN local network is listed in “Asterisk SIP Settings” for NAT’d connections to make external calls (even though in this case we get no audio back to the remote phone for internal calls either.) Does anyone know what might have changed in FreePBX 14 that would prevent the Grandstream GXP21xx phones from working remotely over an OpenVPN connection?

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@kolt wrote:

I planning to make a small phone system using Cisco IP phones. I’ve seen the Endpoint Manager used to create these templates but the free version only supports one brand of phone and buying a license for it is very expensive. Is it possible to setup a phone without having to go through Endpoint Manager and being forced to by a ‘Sangoma’ phone?

Also I just set up FreePBX and make an extension. I tried getting the X-Lite softphone to connect to the profile but it gives me an SIP error 408. User ID is the extension, domain is the server, password is the secret and the authorization name is the extension.

Thank you

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@prozak wrote:

Hello all.

I would like some help with an issue im facing.
I have a private vpn voip network with some remote voip phones to provide support.
Thing is i have made a custom context but ever since when a client calls, i dont get caller id but instead extension number.
If i use the default context from-internal everything works.

In this context any 3digit dialed number leads the extensions defined beliw
I attach my context

[from-internal-villa]
exten => _XXX,1,dial(sip/304,25,Ttm)
exten => _XXX,1,dial(sip/303,25,Ttm)
exten => _XXX,1,dial(sip/302,25,Ttm)
exten => _XXX,1,dial(sip/305,25,Ttm)
exten => _XXX,n,dial(DAHDI/g0/69mynumber,25)
exten => _XXX,n,busy()
exten => _XXX,n,hangup()

Is there anything wrong in the above syntax?

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@gerbelhunter wrote:

I have a working system, with 1 x Sangoma A104d installed. In the Asterisk CLI, when I run pri show spans, I get the following;
*CLI> pri show spans
PRI span 1/0: In Alarm, Down, Active
PRI span 2/0: In Alarm, Down, Active
PRI span 3/0: In Alarm, Down, Active
PRI span 4/0: In Alarm, Down, Active
dahdi show status, I get the following;
*CLI> dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
wanpipe1 card 0 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe3 card 2 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe4 card 3 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)

Now I’ve added a Sangoma A102, which shows up in wantouter status;
~]# wanrouter status

Devices currently active:
wanpipe1 wanpipe2 wanpipe3 wanpipe4 wanpipe5 wanpipe6

Wanpipe Config:

Device name | Protocol Map | Adapter | IRQ | Slot/IO | If’s | CLK | Baud rate |
wanpipe6 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 36 | 4 | 1 | N/A | 0 |
wanpipe5 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 36 | 4 | 1 | N/A | 0 |
wanpipe4 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 35 | 4 | 1 | N/A | 0 |
wanpipe3 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 35 | 4 | 1 | N/A | 0 |
wanpipe2 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 35 | 4 | 1 | N/A | 0 |
wanpipe1 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 35 | 4 | 1 | N/A | 0 |

Wanrouter Status:

Device name | Protocol | Station | Status |
wanpipe6 | AFT TE1 | N/A | Connected |
wanpipe5 | AFT TE1 | N/A | Connected |
wanpipe4 | AFT TE1 | N/A | Disconnected |
wanpipe3 | AFT TE1 | N/A | Disconnected |
wanpipe2 | AFT TE1 | N/A | Disconnected |
wanpipe1 | AFT TE1 | N/A | Disconnected |

and also in Asterisk CLI with command dahdi show status;
*CLI> dahdi show status
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
wanpipe1 card 0 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe3 card 2 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe4 card 3 RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe5 card 4 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
wanpipe6 card 5 OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
However, if I run pri show spans, I can’t see the additional 2 spans;
*CLI> pri show spans
PRI span 1/0: In Alarm, Down, Active
PRI span 2/0: In Alarm, Down, Active
PRI span 3/0: In Alarm, Down, Active
PRI span 4/0: In Alarm, Down, Active

Is there something I’m missing? I’ve run a fwconsole and wanrouter restart.

Thanks

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@SiP1701 wrote:

We are using Yealink T28’s with a FreePBX.
When a user presses the DND button, the phone enters DND but the recording that is played is the generic system recording and not the users unavailable message. Is there a setting I need to set for DND?

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@normic wrote:

Hi there,
our users are setting their speeddials and BLF keys on their phones by using UCP.

But how could they print it?

We’re coming from Gemeinschaft, there has been a nice option to simply print out a PDF with all the given names.

Is this option hidden somewhere?

Thank you.

Regards,
Michael

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@telecomup wrote:

I have a client which is in need of using a random caller ID from a list of caller IDs. I found an article here vicidial-tutorials-and-self-help-articles/set-random-caller-id-in-vicidial-and-goautodial/

But I can’t seem to get it to work on FreePBX. I don’t mess with the contexts.

I tried the following in the extensions_custom.conf
[macro-dialout-trunk-predial-hook]
exten => s,1,AGI(randomcid1.php)
exten => s,2,Dial(PJSIP/voiptrunk/${EXTEN:1})
exten => s,3,Hangup

My trunk name is voiptrunk. But the caller ID never gets changed. The calls come up as private number. In the logs, here is a snippet.

– Executing [s@macro-dialout-trunk-predial-hook:1] AGI(“PJSIP/100-00000015”, “randomcid1.php”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/randomcid1.php
– <PJSIP/100-00000015>AGI Script randomcid1.php completed, returning 0
– Executing [s@macro-dialout-trunk-predial-hook:2] Dial(“PJSIP/100-00000015”, “PJSIP/”) in new stack
[2018-07-10 15:49:52] WARNING[30383][C-0000000b]: app_dial.c:2467 dial_exec_full: Dial argument takes format (technology/resource)
== Spawn extension (macro-dialout-trunk-predial-hook, s, 2) exited non-zero on ‘PJSIP/100-00000015’ in macro ‘dialout-trunk-predial-hook’
== Spawn extension (macro-dialout-trunk, s, 18) exited non-zero on ‘PJSIP/100-00000015’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 14014377851, 6) exited non-zero on ‘PJSIP/100-00000015’
– Executing [h@from-internal:1] Macro(“PJSIP/100-00000015”, “hangupcall”) in new stack

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@hcoello wrote:

Good Morning,

I need to setup the MAX_Contacts for a SIP Trunk; checking the settings on the web is missing;

I have a VEGA100 that show error requesting access
registrar_on_rx_request: AOR ‘VEGA100_WITH_PJSIP’ has no configured max_contacts. Endpoint ‘VEGA100_WITH_PJSIP’ unable to register

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@edisoninfo wrote:

Working system for years. v13.18.0 but main screen says 13.0.195.4
Dialing *xxx usually sends the call straight to the user’s voicemail message. Now, some of them work but others give an error "Your call can not be completed as dialed’. Yes, these extensions all have voicemail. If you call the extension it rings and works fine, but dialing *xxx fails.

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@fearx wrote:

Hello!
I’m facing a different problem: My webphone has finally connected, they have managed to communicate but only one side can talk (usually who initiated the call).

This is the CLI log while the call is active:

0x7f7ba46dd480 – Strict RTP learning after remote address set to: 177.129.4.62:42934
– PJSIP/101-000000a5 answered PJSIP/100-000000a4
0x7f7ba46e2a70 – Strict RTP learning after remote address set to: 177.129.4.62:60897
– Channel PJSIP/101-000000a5 joined ‘simple_bridge’ basic-bridge <5f530cc2-438b-400d-b1de-bd3130bd2ba9>
– Channel PJSIP/100-000000a4 joined ‘simple_bridge’ basic-bridge <5f530cc2-438b-400d-b1de-bd3130bd2ba9>
0x7f7ba46dd480 – Strict RTP learning after ICE completion
0x7f7ba46e2a70 – Strict RTP learning after ICE completion
0x7f7ba46dd480 – Strict RTP qualifying stream type: audio
0x7f7ba46e2a70 – Strict RTP switching to RTP target address 192.168.1.8:60897 as source
0x7f7ba46dd480 – Strict RTP switching source address to 192.168.1.8:42934
0x7f7ba46dd480 – Strict RTP learning complete - Locking on source address 192.168.1.8:42934
0x7f7ba46e2a70 – Strict RTP learning complete - Locking on source address 192.168.1.8:60897

(PS: I would like to know what files I can be posting for a better understanding of the problem. I read the official wiki, it seems that one way audio is one of the main problems when using FreePBX, right? I followed all the steps described there and still the error continues)

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@fearx wrote:

Any tips on this? I read the official Wiki about generating a .csv file (I enabled the function, but the Master.csv file was not generated).

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@glew wrote:

Hi i have a problem with the Dahdi module every time the pbx restart.

when the pbx restart or reboot and i try to make a call say al circuits are busy now.

i need to connect via puty and excecute this steps.

service dahdi start
systemctl status dahdi (to check if the dahdi start)
fwconsole restart

After that i can make and receive calls.

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@namos729 wrote:

Hello Everyone,

I’m hoping maybe someone with more experience can help me out here. I have my google voice number forwarded to my callcentric number that is trucked to my FreePBX system. It works perfectly except that when a caller calls the GV number they dont hear the announcement that informs them that the call is being recorded. However, if you call my callcentric number, you do hear the announcement. Considering that both calls come from the same trunk and follow the same incoming route, I can’t figure out why only those that call the callcentric number directly hear the announcement.

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