similar to: Making a call on Asterisk... new thread or not?

Hello, Does anyone know the developer(s) of the app_queue.so application? I'm looking for the PauseQueueMember and UnpauseQueueMember features of this application for the open source version that only seem to be available on the business edition of Asterisk. Thank You, Timothy Karl tkarl@imminc.com

When someone calls in on a zap channel with FXO and presses an extension, and another user picks up using (*8) I changed it to 888, after a few minutes ( I think 2), the call gets dissconected. The users all use Cisco 7960. I didn't yet have a chance to test it when not using Call Pickup (*8)888. Please help. Here is the screen shot in asterisk: +++++++++++++++++++++++++++++++++++++++

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4

Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call:

I'm a newbie to Asterisk, but I'm moderately knowledgeable about phone systems. Right now, I'm most certainly confused. I have a TDM-04B (four FXO) and four analog FXO lines running into it from an AdTran 616. I have Asterisk working internally, although I could use some help getting incoming calls to answer properly and configuring my outbound dialplan. Here's where I'm

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

I took your suggestion and turned my (ill advised) sudoers bash script into an expect script! It works a lot better this way and is more secure. Because I'm not trying to store a password in a script (which I recognize as a bad idea anyway, I I think I've learned my lesson here). It really works well. But the only thing I'm still trying to figure out is how to put a if statement in

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip extension / sip softphone (linphone) 5. 1 x 800Mhz Asterisk + Linux server 6. Asterisk version is 1.6.2.13 7. 1 x IAX2 incoming trunk from phone provider for 1

I don''t know whether this is a bug or feature, and I don''t know whether this belongs to Gmailer (http://rubyforge.org/projects/gmailutils) in specific or Rails/Ruby in general. I have an instance method in an ActiveRecord model, that upon a web request using WEBrick, spawns off a thread, in which it invokes Gmailer to perform some processing. What fails is that simply

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI> -- Hungup 'IAX2/99999-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations first inside asterisk: > sip show peers Something wasn't

OK, so let me check I understand how the timing must work: 0) New user joins the jukebox and wants to start listening to music. Jukebox app selects a mountpoint name, user ID and password 1) Modify the icecast.conf file with the new mountpoint 2) Send HUP to icecast 3) Set up username:password for new stream 4) Create new stream's ices.conf file 5) Spawn new instance of ices to stream

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following