similar to: bridging <ZOMBIE> ?

I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. #> thread apply all bt ........ ........ Thread 6 (process 20135): #0

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

I was able to get app_prepaid working, but unfortunately I am getting one way audio on the phone that I was placing the call from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

hello, Im tryin to make Calls from MS Netmeeting(h323) to Xlite(SIP) it rings, but as soon as i answered it dissconnects!!!! This is what i get from the Asterisk console: -- Executing Dial("OH323/R27469", "SIP/xlite1|10") in new stack Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265 create_addr: Setting NAT on RTP to 0 Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500 sip_call:

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

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

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

What an impressively zombified thread. Though wondering how 53 bits were supposed to fit into 32 might just warrant revivification. -- Sent from my phone. Please excuse my brevity. On July 20, 2017 5:33:34 AM PDT, peter dalgaard <pdalgd at gmail.com> wrote: > >> On 10 Jan 2013, at 15:56 , S Ellison <S.Ellison at lgcgroup.com> wrote: >> >> >> >>>

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/dd10d5ef/attachment.htm -------------- next part -------------- Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes

Hello, I try to place a call between CISCO IP phone 7905 (h323) and an analogue telephone on DLINK DG-104S (mgcp). Call setup is handled ok - both way, but after connection of the call I get both site no-audio, even codec are Set to alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55

Hello, i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line BRI, setup to point-to-multipoint. working fine, but when both channel on interfaces are used and somebody wants to call in - the response is "ha-la-li" - sounds like msn or called number does not exists. i think that right response should be - line busy tone... right ? can anybody tell me what is wrong ?