search for: strictrtp

Hi all I've got an issue where when I call a number that just plays early media back to me. Instead of hearing the full sequence of tones I hear a short ringing then part of the sequence. What seems odd is that I can see the telephone-event/8000 being passed up the chain but when it gets to Asterisk, it is never sent back to the phone. Instead I just see the usual RTP flows. I've been

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 documentation the values "no" and "seqno" are >> more permissive in allowing other sources rather than less, but I thought >> I'd check. >> > > Setting it to "no" di...

On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering >0xhexnumber -- Probation passed - setting RTP source address to

My setup using 11.25.1 was working. When I installed 11.25.2 I now get "sort of" working. I am using NAT in the setup. When I have an internal phone and call out I get audio both ways. But when I call IN my phone rings but I have no audio. Is there a new setting I need to tweek ? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

...: > 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 documentation the values "no" and >>> "seqno" are more permissive in allowing other sources rather than less, but >>> I thought I'd check. >>> >> >> Setting...

...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 documentation the values "no" and >>>> "seqno" are more permissive in allowing other sources rather than less, but >>>> I thought I'd check. >>>> >>&...

...angoma.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 documentation the values "no" and >>>>> "seqno" are more permissive in allowing other sources rather than less, but >>>>> I thought I'd check. >>>&g...

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

Hello everyone! Philippe, you told me to make a bugreport. Well, here it comes, I'm still not sure, if tis is a bug or a miss-configuration. So I've put up a collection of configurations/output/debug files from a simple asterisk session testing the gtalk call. You can download it here: http://juliencoder.de/ap.txt Or I can mail it, just tell me where and I'll attach it to

Hello, all. We are having a problem where audio for sip channels is dropping upon reinvite. Perhaps it reflects a misunderstanding of what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default ranges of the soft phones (Twinkle on Linux, 3CX on Windows). Testing revealed no problems when the soft phones we used for testing were on the same physical and logical network....

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone --> Avaya Ipoffice --> Asterisk RTP: Phone --> Asterisk But, asterisk send RTP stream to the IPOFFICE and not to phone directly. H323: Asterisk --> Avaya ipoffice --> phone RTP: Ast...

...burn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) Improvements made in this release: ----------------------------------- * ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh) * ASTERISK-24798 - Documentation - Clarify That Form...

...burn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) Improvements made in this release: ----------------------------------- * ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh) * ASTERISK-24798 - Documentation - Clarify That Form...

...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 documentation the values "no" and >>>>>> "seqno" are more permissive in allowing other sources rather than less, but >>>>>> I thought I'd check. &g...

...Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" Containing I...

...Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) Improvements made in this release: ----------------------------------- * ASTERISK-2875...

...Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" Containing I...

...Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) Improvements made in this release: ----------------------------------- * ASTERISK-2875...

...Klaus-Peter Junghanns Posted On Last Updated On August 30, 2017 Advisory Contact Joshua Colp <jcolp AT digium DOT com> CVE Name Description The "strictrtp" option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not originate from the expected address. This option is enabled by default in...

Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to