similar to: Force SIP hang up.

does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set("Zap/49-1", "DID=1231234*4812*") in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hi All; Anyone can advise for a good IP Phone that has the ability to support SIP firmware and IAX firmware? Ofcourse, SIP there is a lot, but we need also the ability to use IAX (as it is good for NAT). Any advise. Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo!

Is there really no way to have asterisk re-resolve domain names from iax or sip providers if this failed or timed out the first time? When asterisk boots on every box i have asterisk is toooo impatient trrying to resolve the domain names for a first time. This results in asterisk thinking the provider is unreachable and only trying again in one week or so. This results (depending on the dial

At Microsoft, we believe that we gain a competitive advantage by making the Visual Studio versioning story as complicated as possible. To wit: The compiler in the first VS 2015 release was version 19.00. For each update/hotfix release, we bumped that version by .01. When VS 2017 was released, we decided to keep the major compiler version the same to signify backwards-binary-compatibility with the

Has anyone had experience using a SIP trunk provided by Paetec over MPLS? With or without FreePBX Regards, Jared Baxley -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120926/436ee928/attachment.htm>

Hi List, While I know that "upping" ulimit will fix the issue I am trying to understand what will cause it. I have a few set ups that are almost exactly the same yet some machines used to give this error often and others don't. I also noticed the error a lot more on my boxes running 1.4.X. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Hi, Is it possible to get the remote channelname that will be bridged when the call is answered, only having the channel that is in the Ring(ing) state? As far as I can see no variable seems to fit when doing the show channel command. I want to be able to redirect/manipulate an incoming call before it gets answered/bridged, but to do that I have to now which channel to use. Is there a way?

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

any way the you can get * to send a NOTIFY SIP message to all SIP phones? to have the SIP sets recheck thier configs etc?? Like this? NOTIFY sip:sip@192.168.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.0/UDP ipaddress From: <sip:webadim@192.168.0.1> To: <sip:sip@192.168.0.3> Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: test@192.168.0.1

Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - hunger at sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 D?sseldorf HRB

How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no messages related to DTMF... or if I just do a global SIP debug for that matter.... I am using RFC DTMF but it's not being passed to the PSTN and I need to debug this further. I've tried to increase the verbosity and the debug ('set debug n') and that didn't help either. I assume this is because even

I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number

Hi, If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646)", does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any

Hi, I have register a sip user to sip server. I can see after registration * is sending periodic "SIP Request: OPTIONS" messages to server. but it's not getting back any response that should be SIP 200/OK as the documents say. 3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip: sipserver.net 3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:

Hi, Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing high CPU utilization from the chan_sip module. I've notice the more sip peers I have loaded, the higher the CPU load goes when there are no active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4 Kernel 2.6.9-42.0.2.EL. I currently have 1558 sip peers loaded in Asterisk and the current CPU load is

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who