similar to: requirecalltoken and Realtime

Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax

Hi i have a small problems on two Asterisk Server 1.6.4 : The first sent the call to the second, and in the second, i have a error : [Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address IP_FIRST_ASTERISK in the calltokenignore list or setting user 04TELNUMBER requirecalltoken=no on the second

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken

> On Jan 2, 2024, at 23:17, thelma at sys-concept.com wrote: > > On 1/2/24 15:13, asterisk at phreaknet.org wrote: >>> On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: >>> I'm using asterisk-16.30.1 >>> >>> When I try to call another asterisk server over IAX I get a busy signal, >>> >>> chan_iax2.c:4739 __auto_congest:

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in

I had two asterisk servers connected with each other, both were 1.4.22 but I've upgraded one to 1.4.37 and now I get a message when I try call from asterisk-1.4.22 to asterisk-1.4.37 ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.140.1 in the calltokenoptional list or setting user guest

Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

    I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them.  Calls come and go but there is no CallerID from the remote server either way.  One of the servers is running Asterisk 16 and the other is an older 1.8 install (I know, I am trying to get permission to update).  The trunk between servers is very simple.  Something like: Server 1 (Mexico) [panama]

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even

Hi, I recently upgraded to Asterisk 1.4.1 from Asterisk 1.4.0 and suddenly my early B3 seems to have stopped working. When I call a number that is out of service from my SIP phones, instead of hearing the announcement from the telco, I immediately receive a SIP Busy message and the call is disconnected. hear no audio. This was working fine in Asterisk 1.4.0 with exactly the same

Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early

I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetwork at gmail.com if you can help. -------------- next part --------------

Dear All, I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. Please help. Asterisk previous version - 1.4.26.1 ( for this worked fine) FreePBX version - freepbx-2.5.2 -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -------------- next part

On 1/2/2024 4:03 PM, thelma at sys-concept.com wrote: > I'm using asterisk-16.30.1 > > When I try to call another asterisk server over IAX I get a busy signal, > > chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow > response >     -- IAX2/192.168.143.1:4569-656 is circuit-busy > > Asterisk-16.16 is working normally, no congestion error. There is not