similar to: PJSIP_DIAL_CONTACTS and Queues

Dear 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

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 offline endpoint. The dialplan

>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to

I have tried both by hand and hitting tab to auto complete: *CLI> channel request hangup Message/ast_msg_queue Message/ast_msg_queue is not a known channel On 31/10/19 14:18, Sean Bright wrote: > On 10/31/2019 2:13 PM, Carlos Chavez wrote: >> I assume this is something created by Freepbx.  If I do a "channel >> request hangup" it tells me the channel does not exist.

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

    Does anyone know if Chanspy can be used with local channels? Since agents on queues need to use local channels like Local/XXXX at from-queue/n to make sure that all of their registered extensions ring we are now having a problem trying to use ChanSpy to listen to calls.  Since we do not know if the agent is on their desk phone or a softphone (which use different identifiers) we cannot set

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 =>

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?

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

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

Hello useRs (please don't kill me), I've fairly new to R having only a few months of playing around with R. What little I've learned has been extremely useful. If someone could point me as to how to replicate the Boltzman Sigmodial curve fit as provided by Graphpad software I'd be eternally grateful. Where we currently use Graphpad for only this one function,its seems highly

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

    I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them.  Calls come and go but there is no CallerID from the remote server either way.  One of the servers is running Asterisk 16 and the other is an older 1.8 install (I know, I am trying to get permission to update).  The trunk between servers is very simple.  Something like: Server 1 (Mexico) [panama]

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

    Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one       s                  59 Up      Dial         PJSIP/1218/sip:1218 at 192.1 17:24:07     I assume this is something created by Freepbx.  If I do a "channel request hangup" it tells me the channel does not exist. Any ideas? --

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls

(Including a bunch more emails in the To: that got missed the first time) Hello everyone! The X.org board is soliciting proposals to host XDC in 2021. Since XDC2020 is being held virtually this year, we've decided to host in either North America or Europe. However, the board is open to other locations, especially if there's an interesting co-location with another conference. Of course

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: > I've split this thread off from another (PJSIP authentication) because I think the root cause is something different. I think the problem is the following FROM line in my SIP INVITE transaction:

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

    I am having a strange problem with a new provider.  We already have a couple SIP trunks working fine.  We are trying a new provider but we are having one way audio problems with outgoing calls.  Incoming calls do have two way audio, only outgoing calls have this problem.  I do not see anything odd with a packet capture and using PJSIP history to check.  The provider says that on outgoing