Displaying 20 results from an estimated 3000 matches similar to: "SIREN14 call setup and record/playback"
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2012 Jan 09
1
video mail is not store
Hi,
I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based.
On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through).
Both the client?use H.264 codec with following sdp information:
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2012 Feb 09
2
Help with Codes and Polycom Phones
Hi All,
This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these:
G722
Siren14.24kbps Siren22.32kbps
Siren14.32kbps Siren22.48kbps
Siren14.48kbps Siren22.64kbps
G7221.16kbps
2011 Jan 05
7
Are the Siren7 and Siren14 the G.722 HD voice codecs?
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal across all
other SIP phones that advertise the HD voice codec like Aastra?
3- What is the main difference between the two and is it advisable to run
these over the INTERnet (not INTRAnet)?
Thanks
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2013 Feb 26
0
Issue with .siren14 sound files
I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14.
I downloaded the codecs and now it will properly transcode to connect
to other phones and play any files that are in .wav format. But when it
tries to play any files with .siren14 extensions, I get complete noise
coming out.
Here's the negotiated SDP:
v=0
o=root 1668560220 1668560220 IN IP4 207.10.184.50
s=Asterisk
2010 Aug 09
0
[SIP/H.264] Codec negotiation problem ?
Hi,
I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : "every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated"
What I observe :
- a call made from a SIP Phone registred in Asterisk to Tandberg works
(voice and video bidirectionnal)
- a call
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my
2014 Dec 11
0
PJSIP configuration question
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP ---
<--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2015 Aug 10
2
Siren7 for Asterisk 13.5
> A Siren codec is not currently available and the one for 12 will not
> work. I have no timeframe for when this might change.
So the only option is to build one from the Polycom sources? I'm
already doing this for Siren14 (I forget why).
2017 Apr 21
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
you mean while placing a video call ? What info am I looking for in the
debug output ?
Kind regards.
J.
On 21-04-17 12:28, Marcelo Terres wrote:
> Did you try to activate DEBUG and set the verbosity to a higher level
> (100?) to check what Asterisk tells you about?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at
2011 Sep 30
1
Core show translation > 4000ms
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the Lenny server
Translation times between formats (in microseconds) for one
second of data
2011 Mar 02
0
Asterisk 1.6 and windows RTC
Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the "peer" section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from windows RTC:
v=0.
o=- 0 0 IN IP4 172.31.9.130.
s=session.
c=IN IP4 172.31.9.130.
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000
2013 Feb 28
1
Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
seems odd that there's no transcoder directly to slin32 since anything
else will lower fidelity. But, more importantly, there is transcoding
from siren14 to slin16 and
2014 Jan 23
1
mixmonitor extension
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---------------------------------------
Marek Cervenka
=======================================
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk