search for: sip_cause

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 => h,n,NoOp(sip cause = ${H...

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 message. So how...

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 this feature by default in future 1.8 versions. This would be an...

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 - logotype] Marcus Kvarsell phone: +46766350384 e-mail: marcus at fogwise.se url: http://www.fogwise.se Like us o...

...ble on the Asterisk blog: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ The following is a sample of the issues resolved in this release: * Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected beha...

...ble on the Asterisk blog: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ The following is a sample of the issues resolved in this release: * Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected beha...

I need to access number received after a I dial a SIP or H323 call? suppose I get one of these: *404 Not found **486 Busy here **408 Request Timeout **480 Temporarily unavailable **480 Temporarily unavailable **403 Forbidden (+) ** 410 Gone **301 Moved Permanently **410 Gone ** 404 Not Found (=) **502 Bad Gateway **484 Address incomplete* How do I get the 404, 486, etc. F.A. -------------- next

...ence/non_digium_state_change --- SKIPPED --> Dependency: sipp -- Met: True --> tests/channels/SIP/custom_info --- FAILED --> tests/channels/SIP/hangupcause --- SKIPPED --> Dependency: twisted -- Met: True --> Dependency: starpy -- Met: True --> tests/channels/SIP/sip_cause --- PASSED --> tests/channels/SIP/invite_no_totag --- PASSED --> tests/channels/SIP/sip2cause --- PASSED --> tests/channels/SIP/sip_outbound_proxy --- PASSED --> tests/channels/SIP/subscribe --- PASSED --> tests/channels/SIP/rfc2833_dtmf_detect --- SKIPPED --> Dependency: ba...

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

...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 ${HASH(SIP_CAUSE,${CDR(dstchannel)})} give the information ? We also face a strange behavior: we are ringing few phones (~10) and sometimes, once the call get answered, we see that 2~3 seconds after this, music on hold is started on the channel! And 20 seconds after, the call is terminated without that any part...

Hello, I'm trying to fetch outbound SIP PROGRESS Reason cause code in the dialplan, Asterisk 1.8.26.1 sip show settings: Q.850 Reason header: Yes Store SIP_CAUSE: Yes However, i'm not getting any value in the dialplan variables, any successful users of this feature? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140430/5f712542/attachment.html&gt...

...PJSIP > (Carlos Chavez) > 2. Re: make asterisk do something when an outgoing call is > picked up (lee) > 3. PlayTones not working (Henry Fernandes) > 4. Re: Register multiple phones to a single AOR with PJSIP > (Scott Griepentrog) > 5. Re: ${HASH(SIP_CAUSE,<channel-name>)} (Paul Belanger) > 6. Re: make asterisk do something when an outgoing call is > picked up (John Kiniston) > 7. Re: AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben > Klang) (Paul Albrecht) > 8. Re: AstriDevCon 2014: Agenda item Dep...

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to determine the real cause of failure. Also, there's no way to link between the channel that was

----- 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 lists...

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

...s defined in RFC 3398. > > * if you think the mapping is wrong, then you should open a ticket on the > Asterisk bug tracker > > * the mapping can only be changed in the code - which you ahve > > * 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, "Reason: Q.850;cause=<cause code>". > > Philipp > > > > > ------------------------------ > > Message:...

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten