similar to: SIP Q.850 Cause

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

Greetings, Recently a performance regression in chan_sip was discovered in Asterisk 1.8. The regression is caused by chan_sip setting MASTER_CHANNEL(HASH(SIP_CAUSE,<chan name>)) after each response received on a channel. That feature has been made optional in the latest 1.8 SVN code, but is currently still enabled by default. After some internal discussion, we decided to consider disabling

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hello, on previous versions of asterisk, extension h and H make us know who ended a call (caller or callee). In the last * versions, seems that only h extension is used, as stated here http://www.voip-info.org/wiki/view/Asterisk+standard+extensions In the last versions, how do we know which end terminate a call (SIP, ISDN, Analog, ...) in h extension ? Will the

----- Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)} Asterisk 1.8 also comes with a 'use_q850_reason' configuration option = for generating and parsing, if available:=20 ----- That will give you what you want if you consider upgrading to v1.8. =09 -----Original Message----- From: asterisk-users-bounces at

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

Hi, I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/ When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message: Unable to

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hello, I'm trying to build a page system using a Dell Desktop PC optiplex 170L, My sound card is working fine under /dev/snd/ exten => s,1,Dial(Console/snd/,20,A(trek)) exten => s,2,Hangup But won't work! I get the following error [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could not find <snd> [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:850 oss_request:

Hello. Calls cancelled by caller during the dialing phase, are shown in Linphone as simply past calls, not missed ones. I thought this is an Linphone issue, but Sylvain says it's on my PBX side: https://github.com/BelledonneCommunications/linphone-android/issues/832#issuecomment-557474864 > The CANCEL message has a Reason header with Q.850 protocol and cause 0, which doesn't mean

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

Hi, I would like to know if someone can suggest me an efficient way of writing a dialplan to match "variable-length" international calls when using SIP clients with the "early dial" or 484 feature. What I usually do for clients that do NOT "early dial" is define something like this in my outbound context: For local calls (they fortunately have a fixed length):