@johnjces wrote:
Hopefully not too stupid of a question but can I run this conversion tool over the Internet on a remote system or must the New and Donor have to be on the local network?
Read the Wiki and other stuff but no mention, and simply curious as I have a couple old boxes to newer boxes needing configuration. One is a 2.11 FreePBX, (if I remember correctly, or I’m remembering old DOS 2.11), which is remote.
Thanks!
John
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Participants: 1
@jeff.wong wrote:
Hi all,
I have just upgraded my system from freePBX 13 to 14, and I able to add any of the widgets in UCP but except FAX, I have tried disable and enable again Fax Configuration Professional 14.0.1.10. Any suggestion?
Btw, I was think that if UCP 14 can have a general dashboard by default will be better. I can imagine how confuse when users first time login UCP 14 from the old version even there is a Tour provided, they’ll still look for administrator support.
Posts: 6
Participants: 2
@mhammett wrote:
[2018-06-10 21:47:31] WARNING[18226]: chan_iax2.c:5005 handle_call_token: Too much delay in IAX2 calltoken timestamp from address
I ran the conversion tool to move a 2.11 PIAF install to the 7 distro. I have another PIAF install on the same host connecting via IAX. Both set to sync time from the host. However, there’s too much delay.
How can I further troubleshoot this? I set calltokenoptional=0.0.0.0/0.0.0.0 on both PBXes and set requirecalltoken=no on both extension and trunk settings.
Posts: 2
Participants: 2
@charneval wrote:
Hi.
I have a new installation of a FreePbx 14 with 8 ip phone “yealink”
What is the better solution for create a shared contacts book for my phones?
And when I have unanswered calls, have the way to see who has looked for me?
I attend a reply because I didn’t know a tool to achieve this.Thanks
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Participants: 1
@mhammett wrote:
New install of the 7 , updated.
[2018-06-11 13:03:33] WARNING[26552]: res_pjsip_outbound_authenticator_digest.c:178 digest_create_request_with_auth_from_old: Endpoint: ‘XXXX’: Unable to create request with auth. No auth credentials for realm(s) ‘asterisk’ in challenge.
I have IP authentication with my provider and a SIP client. Both are done in PJSIP. Calls won’t pass through. I redirected the incoming route to the phonebook directory to isolate where the problem was and it persisted.
There are no credentials beacuse there are no credentials.
Posts: 1
Participants: 1
@calippo wrote:
I’m using FreePBX 14.0.3.6. I would like to do this:
A caller comes from an IVR, hears an announcement and leaves a message on an extensions Voicemail. After this the caller should continue with a new IVR-Menu to hear other announcements. So after leaving a Voicemail recording the call shall not be hung, but continued with an IVR.
Where and how can I setup the extensions voicemail settings from the FreePBX-GUI, that this works?
Posts: 1
Participants: 1
@mikeaph wrote:
After losing power, when attempting to boot into FreePBX, the following is displayed.
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Participants: 1
@supersource wrote:
I’m trying to create an image that I can deploy with the latest FreePBX. In order to do this so I can uniquely activate each one separately, I need to remove the activation from the master image. Is there a way to do this?
Posts: 1
Participants: 1
@boctroy wrote:
I’m looking to switch from ccm to freepbx. We have many phones setup on our network including vg204s and vg224s using SCCP. I’d rather see if i can configure freepbx to fill in for the ccm instead of re-configuring everything to use SIP. I have installed Chan-SCCP and SCCP-manager and i got 2 test phones to ring each other, so I think I’m good there. My current goal is to see if I can get a test vg204 to work. I’m guessing I need a /tftpboot/template but I can’t seem to find anything in google or here that gives any information.
I’m new into phone configurations, any help or advice would be welcome.
Posts: 1
Participants: 1
@bk6662 wrote:
Hello,
Just installed the latest 64-bit ISO, and then ran the conversion tool to migrate settings to my new platform. Everything went well - no errors. But now that I turned off the old system, I’m not getting service at my extensions. The instructions for the tool did say that trunks have been left at Disabled, to prevent any conflict. I don’t want to enable that until I ensure everything else is working internally. Instructions are very vague though. Do I need to enable trunks in order for Internal communications to work?
Thank you!
Brian
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Participants: 1
@supersource wrote:
I’ve got a IAX2 trunk that connects two FreePBX systems working great. I have unique extensions on each system, however, there is a group of extensions, say 4xx that forwards to 5xx on the other system. I use the custom device for this. For example I have extension 434 that forwards to 534 on other system. In the dial field for the custom device for extension 434 I have local/534@from-internal do that calls are forwarded to the actual physical extension 534 on the other system which actually has a SIP device config on it. However, when queue calls for 434 come in, they are forwarded correctly to 534, but require the recipient to press 1 to accept the call. I’m thinking maybe it has something to do with the call context, but I’m using from-internal on the iax2 trunk so I would think that would work. How do I get rid of the call confirmation requirement?
Posts: 1
Participants: 1
@baloo1986 wrote:
hello i have a problem after upgrade of Modules. When i click on APPLY i becom this error:
exit: 1Unable to continue. died in splice ext-did-0002 s in /var/www/html/admin/libraries/extensions.class.php on line 197
#0 /var/www/html/admin/modules/blacklist/functions.inc.php(58): extensions->splice(‘ext-did-0002’, ‘s’, ‘did’, Object(ext_set))
#1 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): blacklist_hookGet_config(‘asterisk’)
#2 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#3 {main}
Posts: 1
Participants: 1
@jsulmar wrote:
I’m migrating from FreePBX 2.9 to 13 using the Distro Conversion Tool. It seemed to work great, except that my IVRs were not converted. Any advice what might have gone wrong?
Posts: 1
Participants: 1
@ecnorm wrote:
Hi FPBX community! Good day. I am a newbie here. We are testing freepbx for a small office situation with need for maximum 3 lines. plus International calling, toll free inbound (U.S) - basic setup. I am evaluating different SIP trunk providers.
I would like to know who do you use for your SIP provider? We would like to keep costs low without compromising quality, so i would appreciate your feedback / suggestions on which SIP provider you use, and who do you recommend?
Thank you!!
Posts: 1
Participants: 1
@kashifiqb wrote:
Hi all,
One of our telecom servers recent shut down unexpectedly due to power failure. After that when we power it ON, it gives this error: http://prntscr.com/ju95lq and doesn’t proceed further. Can anyone help us with this? I am not a Linux expert and don’t know how to resolve this issue? Do I have to do fresh installation?
Posts: 2
Participants: 2
@Azer wrote:
Hi, we have a Sangoma FreePBX connected to HiPath 3800 via SIP trunk. Trunk status is up, Calls beatween PBXs is successfully. But we have a bit problem is Caller ID name is not working via SIP trunk on both side. Please explain for us how to configure freepbx CID settings detailed. there is extension advanced options Outgoing CID. if you write there digits it’s working, but do not supported character(name). please helps us!
Posts: 2
Participants: 2
@claloano wrote:
I would like to realize a feature that when you call a number and this number is busy you automatically activate an activity that every 3 or 4 or 5 minutes will check if that number is free and give a warning to the extension concerned
Ideas are good too
Thank you
Posts: 1
Participants: 1
@tony722 wrote:
Hi,
Running FreePBX 14.0.3.6.
Wondering how to prepend something like "Xfer: " onto the callerID when blind transferring a call.
There are several reasons but here’s one: so that when a supervisor is part of a ring group, they can tell that the call is being transferred to them specifically, rather than just a general ring group call that they would let go to someone else normally. (Supervisors monitor ring groups, but don’t typically answer them, except when things are really busy.)
I saw this post from 11 years ago, but enough has changed that I don’t see how to apply it. Hopefully there’s an easier way now?
Thanks,
Tony
Posts: 2
Participants: 2
@Christopheric1 wrote:
I am having an issue with a 7975g. We have installed the Endpoint Manager module that has the template and everything for the phone. I have made sure in the tftpboot directory that the SEP_macaddress_ file is there and the phone boots up, gets dhcp, pulls down the time and phone name and buttons, but will not register. I checked the logs within freepbx and it is trying to register as “macaddress:sip” instead of “extension:sip”. Is there any way to change the .xml file so that it sends the right info? I have the correct as the extension with the right password that I put into the extension.
I guess I am at a loss. Here is the error:
Via: SIP/2.0/UDP phoneip:5060;branch=z9hG4bK7d98634fFrom: sip:0023339ca0XX@phoneip;tag=0023339ca072000245edc4c2-714a6ed3
Call-ID: 0023339c-a0720002-aa4b2474-d9767c01@10.60.0.100
Contact: sip:0023339ca0XX@phoneip:5060
Referred-By: sip:0023339ca0XX@phoneip
Refer-To: cid:b774e40e@phoneip
Content-Id: b774e40e@phoneip
phoneip/23
Sent:REGISTER sip:10.60.0.2 SIP/2.0 Cseq:101 REGISTER CallId:0023339c-a0720002-6747dc48-842a4b0e@phoneip
[2018-06-13 12:39:29] VERBOSE[2450] res_pjsip_logger.c: <— Received SIP request (933 bytes) from UDP:phoneip:49312 —>
Via: SIP/2.0/UDP phoneip:5060;branch=z9hG4bK69372839
Call-ID: 0023339c-a0720002-03447f20-da5fe0fd@phoneip
Contact: sip:6501@phoneip:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0023339ca0XX”;+u.sip!devicename.ccm.cisco.com=“SEP0023339CA0XX”;+u.sip!model.ccm.cisco.com=“437”
[2018-06-13 12:39:29] VERBOSE[27606] res_pjsip_logger.c: <— Transmitting SIP response (510 bytes) to UDP:phoneip:49312 —>
Via: SIP/2.0/UDP phoneip:5060;rport=49312;received=phoneip;branch=z9hG4bK69372839
Call-ID: 0023339c-a0720002-03447f20-da5fe0fd@phoneip
[2018-06-13 12:39:33] ERROR[2450] pjproject: sip_transport.c Error processing 2073 bytes packet from UDP phoneip:50935 : PJSIP syntax error exception when parsing ‘Request Line’ header on line 1 col 1:
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Participants: 1
@shivaramkar wrote:
Hi,
I have installed new freepbx , i created new sip extension. But when i try to connect the xlite it is not connecting. can someone help on this.
Regards,
Shivaram
Posts: 1
Participants: 1
