Displaying 20 results from an estimated 10000 matches similar to: "No RTP Engine problem in 1.8.2"
2015 May 21
1
asterisk 13 webrtc
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in
SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
2012 Jan 28
1
process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
Hi All,
I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
But when making A Call from SIP Client, I got cli Warning ... and no call
has been made.
My Sip Client is using lib java peers client http://peers.sourceforge.net/
with standard codec PCMU/PCMA
[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 0 8
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
examples of "interesting" information like ICE result and howto make
"minimal" configuration of pjproject.conf
i.e.
forĀ debugging app_queue.so
core set debug 5 app_queue.so
for debugging RTP
core set debug 10 rtp_engine
core set debug 10 res_rtp_asterisk
rtp set debug on
logger.conf
rtp => debug,verbose(5)
so i mean
in
2016 Feb 10
2
Unexpected termination of the call when pick up (res_pjsip)
Please help find the cause of strange behavior res_pjsip.
Making outgoint call to other sip server (CommuniGatePro), my asterisk
suddenly sends BYE after picking up!
Partial log of an outgoing call with full debug is attached and on web:
http://pastebin.com/tLNCpx4d
No diagnostic messages why asterisk suddenly decided to hangup i don't
found :(
There are suggestions or strong belief
2017 May 30
0
Asterisk 14.5.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 14.5.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2010 Oct 25
4
google voice + asterisk: calls made to GV# processed but weird
Dear all,
First off, I am very new to asterisk so forgive me if any of my
comments or questions seem trivial. Thanks to [this
post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/)
and [this post](http://www.davidvossel.com/?p=28), I have GV set up on
asterisk through jabber.conf and gtalk.conf. I can successfully dial
out from asterisk.
I'm trying to set up an
2017 Nov 14
2
RTCP + Stasis causing high memory consumption
Hello Asterisk list,
I've facing a memory allocation issue that happens occasionally but on a
consistent basis.
The problem happens as follow, suddenly Asterisk starts consuming a lot of
memory, in a rate of more than 1GB per hour. Kernel will eventually kill it
via the OOM killer when memory is really exausted... This situation does
not generate backtrace because Asterisk is responsive
2019 Dec 22
2
res_rtp_asterisk.so problem with minimal (ish) chan-sip based Asterisk
Hi,
For years I've been running a minimal (ish) SIP based Asterisk with
the modules based on chan-sip. For various reasons unrelated to
Asterisk the machine the latest incarnation of this configuration has
been updated to Debian Buster and thus to Asterisk 16. Since this
upgrade I have a dependency problem related to res_rtp_asterisk.so.
So the old config was:
[modules]
autoload=no
load
2012 Mar 20
0
Outgoing trunk is restricted to g.729 but need ulaw
Hi,
I am taking over an asterisk system from another person and having an issue
where a sip trunk is restricting the outgoing codecs to just g.729
I am dialing in from a Cisco 7960. The Invite from the Cisco has the
folowing M line:
m=audio 17022 RTP/AVP 18 0 8 101.
So it is allowing g.729, ulaw and alaw.
Asterisk is tandeming the call out over a SIP trunk
sip.conf tandem trunk config:
2020 Jun 08
0
pjsip extensions rings but call drop on answer
Hi,
I created an IAX2 trunk between my old Asterisk 1.4 server (A) and my
new one with v. 16.10.0 (B).
The trunk seems to be up, and the calls are initiated, eg. an
extension from A can dial an extension in B which rings.
However, as soon as the extension in B answers, the call is terminated.
This is what I see in the console of B:
-- Called PJSIP/4053
-- PJSIP/4053-00000002 is ringing
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2017 Jan 06
3
Issue with handling of 480 DND
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
Stripped down example;
exten = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w)
exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1)
exten = 494XXXXXXXXX,n,Hangup()
.....
exten = 98-BUSY,1,NoOp(Busy)
exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
2017 Oct 03
0
Asterisk 15.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2017 Aug 02
2
Asterisk 15.0.0-beta1 Now Available
The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0.
This beta is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.0.0-beta1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this beta:
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2005 Sep 02
1
Call Return
does * support call return?
i want when the operator transfers a call if the transferee is busy or
doesn't answer the call the call return back to operator again...
this feature may be called:
call return on busy
call return on no answer
Paradise Dove
2018 Oct 09
0
Asterisk 16.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
2012 May 09
1
No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Hi,
I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!...
This is the SDP portion that comes in the INVITE messages of calls
2011 Mar 06
0
Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit.
Hello !
My asterisk log is full of messages like this:
[2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP
2833 Event: 0000009f. Not a DTMF Digit.
[2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP
2833 Event: 0000009f. Not a DTMF Digit.
[2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP
2833 Event: 0000009f. Not a DTMF Digit.
[2011-03-06