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Need some help with IAX trunking. I've got six systems: AsteriskM (main) ___________________|____________________ | | | | | Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5 AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk boxes are using ztdummy for timing, they are all using IAX trunking. My calls come in

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

* include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c: Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more

The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

[Sorry if this arrives more than once. I have sent this twice and it never arrived, despite other messages getting to the list O.K.] ----------- Hello, I would like an incoming caller to be able to choose from the menu options in my extension.conf below. Once They have dialled the appropriate digit, * should call two extensions simultaneously: one SIP phone on this * server, and one over a

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXXXXXX,2,Dial(IAX2/userid:password@217.160.244.186/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_iax2.c:5546 socket_read: Call rejected by 217.160.244.186: No authority found --

Hi, I'm trying to set up a call-back system using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Hello, In my sip.conf I have: ;Register and forward FWD numbers to internal extensions register => FWDNUMBER:PASSWORD@fwd.pulver.com/9500 Which should register Asterisk at FWD and then when any calls are made to FWDNUMBER those calls should be forwarded to extension 9500 as specified in the extensions.conf. What I am getting is it is trying to dial the 9500 (IAX Firefly) client twice when

Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211) [May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded

Hello everybody, This is a problem disturb me for long. I run asterisk Asterisk 1.4.14 and A2Billing 1.3 in the same Debian 4.0 ETCH server. And there is also FoneBridge for TDM over Ethernet with E1 to make as well as receive calls from mobiles or PSTN. And IAX2 trunk runs between Asterisk and A2Billing. For most of the time they really do a good job. But some hours or days later----I mean it

Hi All, I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan: exten => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr) exten => 8355,n,Dial(IAX2/8366,48,tTWwr) (i made that simple to exhibit issue) I got just 1 ring in 8366 extension before it hangup, what i noticed is the total time spent on ringing is 30s that means if i use 12s in the first dial i get 18s left

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing