similar to: Asterisk voicemail problem

Hi all, I have installed a test machine with asterisk in order to try it. I have a problem with capi channel (chan_capi 0.3.4a). When an external call directed to an internal Ip phone is not answered I obtain this warning repeated many times: .... .... Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302

Hey, I'm having some errors whenever I dial out and I can't dial in at all. I'm using NuFone as my provider just so you know. Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: No such device Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set device to input mode Jan 11 17:39:46 WARNING[1771]: app_dial.c:359

Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten => 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial "1234" but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got

ser forwards a sip message with extension 99999996 to asterisk which plays my 'userisoffline' message and hangs up and should stop here but instead asterisk continues to process the match everything extension ._ and dials out which is not what I want... if I change the starting priority of the Dial app to a higher level than 3 asterisk stops after the hangup but then doesn't accept

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: -- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

Hi! I am running an * 0.7.2 on an X86 debian stable 2.4.25 (with backports.org). The HW I am using is Digium's E100P on an HP DL 380. Quite often it crashes, e.g. after a call has finished. Below some logs form the * Console as well as from the /var/log/asterisk/messages (Replaced some stuff with XXX). Any idea what there could be the reason for this segmentaion fault? What other

Hello Users I'm Doing working on Both OpenSER and Asterisk ....... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=>115,1,Park() exten =>115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says " U parked on 701 extension " After When 9001 dial

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

When I call into my Asterisk box via my VoIP line (using gsm codec) and then try to make an outgoing DISA call over PSTN I get the following: [Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot handle frames in gsm format [Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to forward voice or dtmf Obviously, it looks like asterisk is not converting the

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 "Redirecting..." back

Hi! As reported earlier this week, I have problems with a sometimes-crashing Asterisk. In most of the cases safe_asterisk is able to restart it. But sometimes it crashes, so that manual interaction is necessary. The seg-faults and crashes occurs, right after call between a SIP Terminal and a legacy PSTN Terminal (PRI/Euro-ISDN), but only if the SIP Terminal hangs up as first. No problem, if the

Hello everybody, my situation is the following: I have an ISDN telephone connected to a HFC ISDN card on an asterisk server. The asterisk server is behind a NAT, but all the ports (i.e. 5060 and the range specified in rtp.conf) are forwarded to the asterisk machine. I am using the German SIP provider Sipgate.de. The sip commands show that I am registered properly with Sipgate. My problem is

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302