similar to: chan_agent and SIP UA transfers fail

Hey, We've been working on tracking all inbound calls to certain call-centre members and have hit a snag, it seems when a queue delivers a call to an Agent, chan_agent will call Local/whatever but provide no means of associating the call in the queue with the local call. To that end i added in an event called AgentAssociate which looks like this: 'DestChan' =>

Yea, I have faked that with a silent mp3, but to do it right it should also be a config flag in the agent.conf file for each agent, prolly add another arg to each agent definition for the MOH class, & the arg 'none' means don't play music for that agent -----Original Message----- From: James Golovich <james@wwnet.net> To: asterisk-users@lists.digium.com

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a "preackannounce" option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would

Hi, I had Asterisk threads consuming 100% CPU at times since last week. Of course, last week an extra card was installed (we had a 1PRI, a 4PRI was added) so search concentrated on that, but to no avail. Today, I installed DDD on the machine and quickly found out that it was looping because cdr->next->next == cdr in ast_cdr_setapp(). I patched this up with some simple code in

Hi, (I'm not subscribed to ext3-users, please CC: me) Kernel: 2.4.18. I've just converted a 100(ish) MiB ext2 filesystem to ext3 (umount, tune2fs -j, e2fsck, mount) and it seems to be happy, except... As a test, I then filled the filesystem up, lastly copying /usr/bin (as root, so the filesystem became brim full). I then umounted it, and ran e2fsck -n -f /dev/hda6, and got the

Hi! For some time, there is a regression in wine, which disables my Dynatext reader from working. It tries to open it's configuration file but unsuccessfully. The file IS there and former wine versions allowed the program to read it. I was recommended to do a regression tests to find a particular patch but it's too time-expensive for me these days, so I did a full trace and found an

Hi, Let me explain what I'm looking for a solution using asterisk. I have one third party SIP based server (A) and on Asterisk server (B). 1. Extension-1 --> Server A calls Server B. 2. Server B does some processing and calls/sends back to Server A ---> Extension-2 3. SIP session has been established b/w two Extension-1 and Extension-2. Now is there any config that I can do in

I am trying to use asterisk for an shared line gateway. When moving from one phone by placing the call on hold then having a second phone pickup that held call by sending asterisk a replaces header (http://www.ietf.org/rfc/rfc3891.txt) Asterisk does not seem to send a "bye" message to the original UA leaving the first phone stuck in a holding state. Am I missing something here? Here

Good evening list, I'm looking for an R implementation of the "Shuffled Complex Evolution??? (SCE-UA) algorithm after Duan et al. (1993). Does anybody know if there is an extension/ package existing that contains it? Thanks very much for your help! Cheers, Simon Duan QY, Gupta KV, Sorooshian S (1993) Shuffled Complex Evolution Approach for Effective and Efficient Global Minimization. In

Hi, I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine. I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is

I've got the following situation where a UA is trying to call another UA via Asterisk and SER according to UA1 -> * -> SER -> UA2. Now in the event that SER generates a 404 Not Found for UA2 I would like Asterisk to return or relay or forward or whatever the 404 to UA1. Anyone know this might be able to be done (or maybe not possible at all?) Craig

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

Hi I registered my asterisk box to my SIP provider as an UA. For every call I receive on this trunk, I get the message "That is not a valid conference number". I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. Szasz Szabolcs -------------- next part

Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ?externaddr?, ?localnet? and ?nat=force_rport,comedia?. Asterisk registration is successful, I see in Wireshark the packets send between Asterisk and SIP server. However,

Morning! I'm trying to subscribe a softphone to an Asterisk 11 server, but it sends an "ua-profile" event that Asterisk immediately rejects with a 489 Bad Event error. Is this event not supported at all? Are there any workarounds? Best regards, Norman -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

I am using Asterisk CVS-HEAD-06/02/05-19:37:27. It seems that every call I made was duplicate. Jun 8 00:11:30 DEBUG[21733]: chan_h323.c:411 oh323_call: Placing outgoing call to 87874586, 101 -- Called 87874586 Jun 8 00:11:31 DEBUG[21733]: rtp.c:472 ast_rtp_read: RTP NAT: Using address 10.17.43.53:8000 Jun 8 00:11:32 DEBUG[21735]: chan_h323.c:1218 progress: Received ALERT/PROGRESS message