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On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the docume...

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an...

Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explains libpjna...

...remote > address set" should reset the rtp_source_learn.start timestamp, and yet the > "Strict RTP learning complete" messages are less than 5000ms after that. > What could be happening? > > Thanks again. > > > On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp <jcolp at sangoma.com> wrote: > >> It's probably best if you read the logic[1]. There's an entire comment >> that talks about how it works. >> >> [1] >> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 >> >> On Mon, Apr 17, 2...

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> This stems from PJSIP not being dynamic with transports (it JC> doesn't like its environment changed to that degree while JC> in use). I'm afraid if your IP changes you'd have to restart JC> Asterisk when you are using PJSIP. Wow. I say this...

...learning a new audio stream is a minimum or a maximum? The unusual call flow in question results in Asterisk learning a new audio stream when we don't want it to, and having a minimum of say 2 seconds of audio would help avoid this. Thank you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > >> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >> dcunningham at voisonics.com> wrote: >> >>> Hello, >>> >>> Does anyone...

...a minimum or a maximum? The unusual call flow in question results in > Asterisk learning a new audio stream when we don't want it to, and having a > minimum of say 2 seconds of audio would help avoid this. > > Thank you! > > > On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > >> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: >> >>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >>> dcunningham at voisonics.com> wrote: >>> >>>> Hello, >>&g...

---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Sunday, October 4, 2015 12:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:09 PM, Bryant Zimmerman wrote: > -- > Joshua > Thanks for your reply. It thought the same thi...

...econd "Strict RTP learning after remote address set" should reset the rtp_source_learn.start timestamp, and yet the "Strict RTP learning complete" messages are less than 5000ms after that. What could be happening? Thanks again. On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp <jcolp at sangoma.com> wrote: > It's probably best if you read the logic[1]. There's an entire comment > that talks about how it works. > > [1] > https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 > > On Mon, Apr 17, 2023 at 7:10 PM David Cunningham...

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent v...

On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT <lists at telium.io> wrote: > >> That answers part two…but is there any mapping of AMI version to Asterisk >> versions? >> > > No, there is not. > I can say that Asterisk 13 is 2.x.x though beca...

Oh, wow! Changed it and now I am getting calls into my context (fromgw). Unfortunately, the actual caller ID (6175551212) is not getting passed (but I know Asterisk is getting this). How do I "reap" this actual caller ID in my dialplan? On Fri, Mar 13, 2015 at 4:55 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > [sonnyGW1] >> type=identity >> endpoint=sonnyGW1 >> match=65.254.44.194 >> > > You want type=identify, not type=identity. > > Cheers, > > -- > Joshua Colp > Digi...

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames

...can't get any updates? Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board? Thanks Bryant ---------------------------------------- From: "Joshua Colp" <jcolp at digium.com> Sent: Tuesday, January 26, 2016 8:10 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] PJSIP Stun/ICE Bryant Zimmerman wrote: > Joshua > Since there is no automated way curr...

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research...

Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file? It seems to only read it on startup. Thanks.

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Wednesday, May 13, 2015 10:10:25 AM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32...

I mean by sending image by using sip channel just like we can send text message and what about sending image file ? On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp <jcolp at digium.com> wrote: > On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote: > > Dear Sir, > > Kia ora, > > > > > I current have done successfully with sip message over asterisk server , > > and additionally now I want to send the image between sip using asteri...

On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp <jcolp at digium.com> wrote: > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > > This specific issue exists in a lot of different implementations and > devices. Unfortunately there's nothing within SDP that...