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Hi, Is there any way I can get exact sip status from pjsip after a dial ? or all we can get is asterisk hangup causes ? Thanks in advance. KKh

...show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me that we cannot access sip code of a call > from pjsip which information is actually returned from the provider, > so it is available to asterisk, why does asterisk hide it ? > On Sat, Jun 9, 2018 at 5:08 PM Khalil Khamlichi > <khamlichi.khalil at...

Hi, I am looking to configure asterisk queues in off-hook mode, that is, the agent calls into the system and stays connected to this call, when new customer calls, he is redirected to the queue which should distribute to connected agents. is this possible on teh actual app_queue or we would need to implement it using ARI. Thanks in advance.

I *am* doing that, as I assumed it would be required just for the 911 mapping we have provided, but that doesn't change the SIP header. Cheers, j On 05/08/2018 02:41 PM, Khalil Khamlichi wrote: > try setting the callerid with > > same => n,Set(CALLERID(all)=17864089672 <17864089672>) > > ofcourse for each customer you will need to provide his own did. > > > On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere <jeff at stratustalk.com > <mailto:je...

Hi, Is there a way to add another extension to a live dial, for example Dial(PJSIP/1000,,) and after 20 secondes change it to Dial(PJSIP/1000&PJSIP/1001,,) I am open to suggestions such as using manager or stasis. Thanks in advance. Best regards, Kkh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Hi, We have been using Voxbone for some time for origination, and they now offer E911 services.? We are trying to set this up and having trouble meeting their authentication requirements. I setup a peer as I normally would, with user/pass as they supplied ("lacoursj", "pass"), but my calls are rejected.? Their support is asking that I follow this auth mechanism: 1st step