similar to: Error, Clue to what?

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

Hi, I am trying to use asterisk open source version(asterisk-1.6.0.5) with MySQL (using res_odbc)support for extensions and users list. The call rate is 7 calls / second and each call stays for 120 seconds. after making 25000 successful calls , calls started failing with following message on CLI. [Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable to create channel of

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, I follow the web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf - mohstream.sh , to configure music on hold to play using mms but failed. Anyone can play using mms? ango

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

Hi all, I'm new to this list, so forgive me if I'm not supposed to ask this: I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there any way to use TLS with this phone<--->asterisk (v 1.6.0.9)? It is said that is supports TLS/SRTP but I don't see any of these options in the configuration file or the admin (advanced) SIP conf panel. Am I missing something? Thnx

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

Hi few weeks ago I read about redundancy (HA) of asterisk boxes using DNS, DUNDi, so I decided to give it a try. OS FreeBSD 6.1-RELEASE, asterisk 1.2.7.1 on one peer I get: lk110*CLI> dundi show peers EID Host Model AvgTime Status 00:11:43:3d:69:e6 195.28.109.37 (S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0

Greetings, I'm having a heck of a time with one way audio on a SPA2012. It's public IP connected directly to cable modem. One line configured. Asterisk is multihomed Public IP outside / Private Inside. Extensions inside network are can't hear audio from phone outside connected via the spa-2012. Outside can here audio from inside the network. Ring works both ways. I've

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

Hello when I set qualify = yes on trunk I can't do outgoing call. Incoming is always working. [Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but my linphone is registered all the time. when set qualify = no outgoing call is working (but i have problems when WAN IP is changed after

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers.