similar to: Problem ringing simultaneous channels

I just installed a new TDM400P with one FXO interface in slot 4 (how it came from Digium). This box is running Debian with a 2.6.10-rc2-mm3 kernel. After the make linux26 and make install in /usr/local/src/zaptel, I can see contents in /dev/zap but any attemp to touch for example /dev/zap/ctl gets a no such device or address ... Any suggestions?

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see

hi, The prob is when we ring 2 channels simultaneously, only 1 channel is actually ringing. In our configuration, the Asterisk box is connected to an E1 channel bank, where 15 analog extensions are conencted to channelbank inturn. We tried following, Dial,Zap/g4/444&Zap/g4/448|20|t Heres the output, -- Executing Dial("IAX2[trunk10@trunk50]/1",

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

Hi, For some time now I have noticed that our RBS T1 (asterisk 1.4.35, Dahdi 2.3.0+2.3.0, TE410P) often has channels stuck in the state "Ringing", like this poor chap who got stuck on two calls in a row, apparently: [excerpt from "core show channels"] SIP/7157997-0000534b 7760308 at business:1 Ring Dial(Dahdi/g0/7760308) DAHDI/3-1 5130262 at from-pstn:1

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Hrm, All of a sudden for some reason Wait() and Playback() are returning non-zero and its causing calls on my inbound SIP leg not to complete. I'm not sure why -- Executing Answer("SIP/2181-4518", "") in new stack -- Executing Playback("SIP/2181-4518", "silence/1") in new stack -- Playing 'silence/1' (language 'en') == Spawn

Hello, Is it possible to set up a timeout for IAX when something like the following happens? -- Executing Dial("SIP/someone","IAX2/somewhere|45") in new stack -- Called somewhere -- Call accepted by 1.2.3.4 (format ulaw) -- Format for call is ulaw <nothing happens here for 15 - 30 seconds - caller gets tired of listening to silence and hangs up> -- Hungup

I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hi, In my Office all our users have a Desk phone. Some of the users who are using laptops have a Soft phone too along with their Desk phone. Right now we are using two different extensions for their desk and soft phones. Is it possible to have simultaneous ring for both the extensions (ie. soft phone and desk phone). I tried using " Dial(SIP/desk & SIP/soft) " and it works fine

Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a "ring group" and "hunt strategy" How can it be done

Hi; I have entries as below in DB, mysql> select * from sip_buddies; +----+------+----------+------------+---------+------------+--------+------- -----+------------+----------+------+ | id | name | context | defaultip | host | mailbox | type | regseconds | ipaddr | username | port | +----+------+----------+------------+---------+------------+--------+-------

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

I have a client who is looking at replacing their PBX, and I'm interested in putting together an Asterisk solution for them. One feature that would really, really get their attention is if I could do the "Vonage" thing, where if a PSTN caller dials a direct extension (coming in over PRI) both the user's deskset _and_ an external number (their cell phone) would ring, with

I have a need to issue a dial command to a number: same => n,Dial(${DIALGROUP1},${TIMER1},t) After a number of seconds, let's say 10 seconds. I want to dial another set of numbers while continuing to ring, or interrupting the first group of numbers. same => n,Dial(${DIALGROUP2},${TIMER1},t) Is there a way to do this without interrupting the first call? Thanks, Scott Haley If you