Displaying 20 results from an estimated 3000 matches similar to: "Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)"
2023 Jul 05
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello Michael,
you are referring to the following behavior - did I get it correctly?:
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and callee
(g722 <-> g711).
inbound works: call from provider: g711 -> asterisk drops g722 and
passes g711 to internal callee -> no transcoding.
As far as I know,
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
After I have re-read the "PJSIP Advanced Codec negotiation" document, it
occurred to me that the desired behavior should actually happen
automatically, just due to the codec negotiation logic, but it looks
like asterisk doesn't actually follow the described logic which is
likely a bug.
Can you please follow with me through a simple sip call and see if I'm
missing
2023 Jul 05
3
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Oh, that's great. It wasn't clear from that page, at least not for me. :-(
Having it clearly stated on the document would save me (and probably
others) lots of time.
Thanks for clarifying it. Any idea on the timeframe of implementation?
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/6/23 12:47, Joshua
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com>
wrote:
> Hello,
>
> After I have re-read the "PJSIP Advanced Codec negotiation" document, it
> occurred to me that the desired behavior should actually happen
> automatically, just due to the codec negotiation logic, but it looks like
> asterisk doesn't actually follow the described
2023 Jun 30
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on originating
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on
outbound (called) channel and has no affect on calling channel. My
experiments and function documentation (which says “Media and codec
offerings to be set on an outbound SIP
2017 May 12
3
pjsip: asterisk can't decide which codec to use
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone extension -> asterisk -> provider A -> provider B -> asterisk.
Asterisk initially sends
2017 May 13
2
pjsip: asterisk can't decide which codec to use
On 05/12/2017 at 08:49 PM, Joshua Colp wrote:
> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
>
> <snip>
>
>>
>> If I'm doing exactly the same call originated with another extension,
>> there can't be seen these frequent changes. But the strange thing is,
>> that in both cases the part between extension and asterisk doesn't show
2020 Feb 25
0
pjsip startup errors when using "with-ssl" configure option
On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote:
> Hello list,
> Hope you are all doing well!
>
> I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
> I wonder if someone can put some light on it.
> Log history short, install_prereq fails to install the packages (not sure
> how important they actually are....):
2013 Dec 15
3
Why doesn't Asterisk try to prevent transcoding
Let's say I have two devices configured and the follow call scenarios occur.
[100]
disallow=all
allow=g722&ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with g722,ulaw,alaw,g729
101 dials 100 -> ulaw to ulaw is chosen
100 dials 101 -> g722 to ulaw is chosen
Ideally when 100 dials 101 ulaw would be chosen since it is the common
format.
2018 Jun 16
2
Only 8kHz recorded after disallowing all but G722 codec on inbound
We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.
My test calls show inbound to the proxy is recorded at 16kHz, inbound in
Asterisk is only 8kHz, and the peers receive 8kHz. So
2014 Dec 05
0
Yealink/G722/No Outbound Audio?
So I've got a bit of a head scratcher. Wanted to get some insight.
I've got a PBX running 12.3.0
We're a ULAW shop from end to end. But I've been playing with G722 just
for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom
IP650 (Same office).
Basically, Whenever I make an outbound call to a destination to something
not G722 ready, I get no
2020 Feb 21
2
pjsip startup errors when using "with-ssl" configure option
Hello list,
Hope you are all doing well!
I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
I wonder if someone can put some light on it.
Log history short, install_prereq fails to install the packages (not sure
how important they actually are....): speexdsp-devel, gmime-devel,
uriparser-devel, iksemel-devel, uw-imap-devel, hoard
Then, I am running the following commands
2019 Oct 03
2
Asterisk not using common codec between (SIP) endpoints
Hello people,
I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):
In short: - Asterisk behaves unexpectedly (at least to me) when
negotiating between endpoints
that have a different but intersecting set of codecs
(preventing direct media flow).
- Also, when an endpoint sends RTP with an
2020 Feb 25
2
pjsip startup errors when using "with-ssl" configure option
Hi Kevin!
Thanks very much for your reply! Much appreciated!
So I just have a remaining question from this, if the with-ssl is not
mandatory to have the encryption support, what is it actually used for?
Maybe it is some old flag which is not needed anymore and so can be ignored
for now and possibly removed from the configure/makefile stuff for future
releases?
Kind regards,
Patrick Wakano
On
2009 Mar 06
1
Wideband (G722) MeetMe
Hi all, I?ve read that meetme works at G711 (ulaw), so asterisk would
down-mix a number of parties using G722, is that still correct?
If so, i?ve also read that Joshua Colp was/is working on a replacement
(conf_bridge?) that works with G722. If this is this still in active
development are there any planned timelines? If it?s in 1.6.0.6, and i?ve
just missed it or it?s been renamed please be nice
2015 Aug 13
2
simultaneous use of chan_sip/chan_pjsip
hello,
is it possible simultaneously use chan_sip and chan_pjsip?
if yes, can you recommend settings
i'm thinking about
- chan_sip - for sip hardphones/softphones (sip udp 5060)
- chan_pjsip - for webrtc
--
---------------------------------------
Marek Cervenka
=======================================
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
2015 Aug 13
2
simultaneous use of chan_sip/chan_pjsip
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
> On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz
> <mailto:cervajs at fpf.slu.cz>> wrote:
>
> hello,
>
> is it possible simultaneously use chan_sip and chan_pjsip?
>
> if yes, can you recommend settings
>
> i'm thinking about
> - chan_sip - for sip
2003 Jul 16
3
Segmentation fault with chan_oh323
Hi,
I'm trying to interconnect sip and h323 endpoints using asterisk
and asterisk crashes with segmentation fault whenever h323
connection needs to be established. It registers with gatekeeper ok though.
Here are the symptoms.
If the call initiated by SIP device, asterisk replies to it "Trying" and then
silently crashes (it launched as asterisk -vvvvcd).
In debug log I can see the