Archives : March-2023
Home » 2023 (paged 13)
As some of you noticed (and likely some didnt) the mailing lists were down for a period of time at the start of this year, and over the past week. We believe weve stabilized things to allow them to continue to run and will continue to monitor.Somet..
My last post did not make it back or to the archive… testing…
Mit freundlichen Grüssen
-Benoît Panizzon-
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I m p r o W a r e A G-Leiter Comme..
Gang I noticed, that when I enable multiple codecs and rtp encrypting (generating a large SDP) invites with credentials do not get through anymore. So sniffed the connection and found that the IP packets have the dont fragment bit set, causing a V..
Weve recently hit an issue with Asterisk 18.8.0 where a call comes in via SIP (using pjsip) but it can take 5 seconds before starting to execute the dialplan. This was intermittent, but frequent (eg approx half of the calls). We have verbose logg..
We have a system that interoperates with an external service, so that the basic call flow is:PSTN origination -> Asterisk A -> External service -> Asterisk BInitially the SDP from the external service tells the two Asterisks to send RTP directly to e..
Hi. I have a strange problem and Im looking for suggestions on how to investigate it. I have a dialplan which is processing a call, and Asterisk simply stops doing anything for that call. I have verbose and debug logging turned on. There are two st..
I see this in my logs:[Feb9 15:25:27] NOTICE[2959153][C-000006c8] chan_sip.c: Failed to authenticate device ;tag19177874 for INVITE, code = -1[Feb9 15:29:44] NOTICE[2959153][C-000006cd] chan_sip.c: Failed to authenticate device ;tag01847080 for INVI..
all, Curious if the github user mlan is on this list? Could you please contact me off list if so, I was hoping to reference your work in a talk at Astricon next week, and… I dont know how to contact github users lol. Cheers, — Jeff LaCoursiere StratusTa..
Over the weekend, we had several customers running at AWS.AWS had an outage during this time.This customer is running Asterisk 16.23.0 (which has the STUN timeout crash fix). From what I have been told, other customers are running newer Asterisk 18.1..
Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints:[Alice]disallow:all allow:ulaw,alaw,g729[Bob]disallow:all allow:ulaw,alaw,g729Alice calls into Asterisk on ..