search for: pjsip_dial_contacts

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an of...

...ear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with ${PJSIP_DIAL_CONTACTS()} to be able to Dial() all of them simultaneously. But there are also situation where you need to Dial() not only one endpoint, but multiple ones, even mixing technologies like IAX and SIP. You can specify those multiple endpoints with the & separator in the Dial() function. Unfortunately i...

...ser's softphone aor: https://blogs.asterisk.org/2017/11/29/pjsip-mis-configuration-can-cause-loss-sip-registrations/ and the second appears to be an artifact of syntax processing and not trivial to deal with. PJSIP doesn't dial all contacts when dialing an aor, so one needs to use PJSIP_DIAL_CONTACTS. However, that can return the empty string and thus lead to a syntax error, leading to the need to write code to fix formatting: https://asteriskfaqs.org/2019/06/09/asterisk-users/dialpjsip_dial_contactsalice-pjsip_dial_contactsbob-how-not-to-fail-if-one-endpoint-has-no-registered-aor.html...

...ic started many of our customers have begun to move agents off site.  Since most of them were using softphones we did not have any problems but now we have one case where the agents have a desk phone in their cubicle and are using a softphone from home.  For regular calls there is no problem as PJSIP_DIAL_CONTACTS dials all the contacts for the extension, but Queues are only sending calls to a single (the first) contact so Queue calls are going unanswered.  For the moment we will power down the office phones but is there a solution for the Queue to ring both phones? -- Telecomunicaciones Abiertas de Mé...

Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten => s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})},20,TtWw) exten => s,n(afterdial),Goto(s-${DIALSTATUS},1) Granted the particular above example could probably be better written to simply modify $ARG2 based on ${SIP} rather than having two Dial() branches, but using the above as just an exam...

Hay guys, have question. When I do regular dial I use $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); to get all contacts of current endpoint and so I dial to all phones at once, but if I exec QUEUE, I have just one phone rings, seems like it take first one as Dial app by default, is there way to fix this?

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

Ok I've got multiple phone sets registered with the same extension/secret. However, this causes a strange problem. If I have 3 phone sets registered on extension 123, and I then call extension 123 (from extension 456), only a SINGLE phone set will ring. Is this by design or a bug? Does only the most recently registered phone set ring when I call the extension? Seems odd...is there a way

I have an endpoint with multiple phones registered as aor contacts. When I attempt to originate a call it will only ring one of the phones. Is it possible to ring multiple phones as a single endpoint. First phone to answer wins the call and all others terminated? Again, using AMI. Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just finished installing Asterisk 13 on our test server and I can now use PJSIP to register phones and make and receive calls. The only problem I am having is that when I register multiple phones to a single account only one of them rings. The AOR for the account has maxcontacts at 3. If I do a pjsip show endpoints I can see two "Contact" entries which I take to mean that

...te that you took the time, but really looking for an answer to the "better way to branch" question. > My braces and parens may be off in this example sorry if it doesn't > work > out of the box. > exten => s,n,Dial(${IF($["${SIP}" = "PJSIP"]? > ${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})}{ARG2})},20,Tt > Ww) Hah. This won't work either, for the reason I asked about in my followup question about where the "true" value of an ${IF ...}has a colon in it and is taken to be the end of the true value and the start of...

Heya Rodrigo Not sure, but this expansion on Sammy's concept may help you achieve the delayed ring on the secondary extensions you were looking for. exten => _600.,1,Dial(PJSIP/${EXTEN}) exten => _600.,n,Hangup exten => _600.wait5,1,Wait(5) exten => _600.wait5,n,Dial(PJSIP/${EXTEN:0:4}) exten => _600.wait5,n,Hangup exten => 555,1,Dial(LOCAL/6001&LOCAL/6002.wait5)

...phone is getting the calls. It's a mess. This is true for chan_sip. It is not true for the PJSIP stack. The PJSIP stack does allow for multiple devices to register contacts to a single Address of Record (AoR). You can then dial contacts individually, or dial all contacts on an AoR using the PJSIP_DIAL_CONTACTS function. I would say that configuring the PJSIP stack in such a fashion is one of the more "advanced" uses, and there are some gotchyas going with that configuration (mostly related to device state). But it is possible. > It's not that a single SIP registrar or proxy cannot have...

Hello, My company has invested heavily in Counterpath?s Stretto provisioning platform for Mobile and Desktop VoIP clients . At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , which many of our users require. Their provisioning system assumes that both devices will use the same SIP extension for auth however. Normally

...tact is a SIP term, it's a way of getting to something. (IP address+port) > So what happens if one endpoint has multiple AOR's which are registered > from different ip addresses. > And you Dial() that endpoint, will PJSIP send invites to all the ip > addresses? If you use the PJSIP_DIAL_CONTACTS dialplan function a dial string will be produced which calls everything. > Is there any practical use for such a setup? It depends. If you don't need them to be individually addressable then it can be useful. > Also I notice, an AOR does seem do be directly correlated with an auth &g...

Nick Awesome wrote: > Hay guys, have question. > > When I do regular dial I use > $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true); > > to get all contacts of current endpoint and so I dial to all phones > at once, > > but if I exec QUEUE, I have just one phone rings, seems like it take > first one as Dial app by default, is there way to fix this? There is no way to d...

...ontact) is CHANNEL(endpoint) is 9150 In my testing, ${CHANNEL(contact)} is always empty. 1. Can someone show me the output of a successful CHANNEL(contact) ? 2. Suppose Alice and Bob phones are both registered as extension 1000, what is the most efficient way to remove Alice's contact from ${PJSIP_DIAL_CONTACTS(1000)} value if Alice ever dials 1000 (and hopes to ring Bob's phone only) ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200124/687b2ed9/attachment.html>

...ttempt to originate a call it will only ring one of the phones. > > > > Is it possible to ring multiple phones as a single endpoint. First phone > to answer wins the call and all others terminated? > > Again, using AMI. > No, you have to dial a Local channel that then uses PJSIP_DIAL_CONTACTS. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200527/dd9b6b99/attachment.html&g...

Hello everyone, I have started testing the PJSIP stack. I saw that it is possible to setup statically multiple AOR contacts, setup qualify_timeout and attach it to an endpoint, and then dial using this endpoint. When I setup the configuration I used the cli in order to see the status of the contacts, and it worked fine - whenever a contact is unreachable, the status is updated to Unavailable.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more