similar to: Invalid Extension

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

I use Xlite and Asterisk. Now, everything was working fine till yesterday. But when my agent tried to login to asterisk through xlite, I see below line sin CLI : == Manager 'sendcron' logged on from 127.0.0.1 -- Got SIP response 486 "Busy Here" back from 192.168.0.17 > Channel SIP/cc101-08969e60 was never answered. == Manager 'sendcron' logged off from

Vicidial can't call and transfer to my softphone. I get some line that says Spawn Extension....exited on non zero.... Here's some of the CLI output. I am using Asterisk 1.2.4 and astguiclient 1.1.8 ...thanks for the help |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time <

I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not been able to send useful callerid info between them (callerid becomes "serverB"). serverA register statement: (serverB has the exact opposite statement) register => serverA:serverApassword at IP_of_serverB_nic/serverB users.conf of serverA: users.conf of serverB: [serverB] [serverA] type=friend

I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: > /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi > install_driver(mysql) failed: Can't load > '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' > for module DBD::mysql: libmysqlclient.so.15:

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

In the Asterisk log I see: --- [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] <--- SIP read from UDP:198.38.7.34:5065 ---> SIP/2.0 200 OK To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport Call-ID:

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

Dear R-listers, I am fitting bivariate mixed models for cost-effectiveness data of cluster randomized trials using lme in R. So I have individuals nested within clusters. My response variable is a vector with bivariate response (individual level costs and effects) stacked into a single column. The covariates in my models are a constant and a treatment term. They are response-specific, e.g. a

timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? [1112221234] exten => s,1,Ringing exten => s,2,Wait(1) exten => s,3,Answer exten =>

Hi Folks, I'm trying to test out Asterisk overall. I'm having some problems with DTMF. Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the

Hi Daniel, I noticed that the patch "Add support for hotplug/unplug of disk devices in LXC" you wrote had been merged into libvirt of version 1.0.2. But when I used this function, it report an error with details as following: ubuntu@lxc:~$ vir attach-device instance-0000002c disk.xml --config error: Failed to attach device from disk.xml error: Unable to create device

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized