Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
Il 15/01/2014 10.09, James Sharp ha scritto:
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
^
On 1/15/2014 5:50 AM, Gareth Blades wrote:
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
Il 15/01/2014 09.59, Francesco Namuri ha scritto:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
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.
On Wednesday 09 May 2012, Ricardo Carvalho wrote:
[May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!
Any help?
Are you sure you compiled all the codecs you need?
What happens if you run `make menuselect` in both the 1.4 source tree
That's weird, because it's negotiated with success the codec ulaw for
outbound calls through the same SIP trunk.
Besides, ulaw and alaw shows up when i do core show codecs audio in the
asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules
under the path
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No compatible codecs, not accepting this offer! -
after upgrading to 1.8.11
That's weird, because it's negotiated with success the codec ulaw for outbound
calls through the same SIP trunk.
Besides, ulaw and alaw shows up when i
On 5/9/2012 11:56 AM, Ricardo Carvalho wrote:
That's weird, because it's negotiated with success the codec ulaw for
outbound calls through the same SIP trunk.
My guess is the incoming call is not being matched with the peer you are
expecting. Do a sip debug and watch the output to see what
Problem SOLVED.
You'r right, this is a problem of codec mismatching. Activating sip debug i
can see it:
Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d
(g723|ulaw|alaw|g729)
[May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!
I
I'm running asterisk 1.2 with Sipura adapters.
I've tried to experiment with different codes but I'm either getting No
compatible codecs if I use gsm or
static noise if I use g726
I was under impression that asterisk would translate between codecs according
to show translation table.
2.) Does
Resolved!
The problem was with sip.conf file. I had to comment the lines
allow=alaw
allow=ulaw
This made the trick..
I am trying to get the mysql database connection from my asterisk box.
Installed the asterisk-addons-1.4.4 version on the same box where asterisk and
mysql is installed.
I read from the forums, that if i build mysql the problem will be resolved. As
i get the similar warning message for MeetMe().
How to build mysql or MeetMe manually??
Regards,
Naveen.Palani
- Original Message -
From: Naveen Palanimailto:[EMAIL PROTECTED]
To:
of the screen, you should see what is
missing.
Good luck.
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Naveen Palani
Sent: 19 February 2008 02:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No compatible codecs!
I read from the forums, that if i
Naveen Palani wrote:
Hi,
I have the asterisk-1.4.11 set up installation on my Ubuntu machine.
When i try making a simple incoming call using xlite softphone. I get
the following message when i try calling to the number.
*CLI [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp:
Hi,
I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try
making a simple incoming call using xlite softphone. I get the following
message when i try calling to the number.
*CLI [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No
compatible codecs, not
Hello,
I have been having this problem for several releases of Asterisk.
Whenever, I use iLBC or Speex codecs to make a SIP call, I get No
compatible codecs! error, even though I am not disallowing anything
in my sip.conf. One way I made it work is to hard-code these codecs
in
global_capability
Hi Eric!
and do NOT use bandwidth=
Why is that? I am curious...
Cheers, Philipp
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Philipp von Klitzing wrote:
Hi Eric!
and do NOT use bandwidth=
Why is that? I am curious...
Because bandwidth= just enables specific codecs. The specific codecs
enabled depend on the bandwidth= setting.
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Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0
sources that I downloaded from Digium, my Polycom 300 works just fine.
If I build with either various CVS builds, or the 1.0.6 sources from
Digium, I get No compatible codecs!.
WTF?
I'm using -the exact same- config
Ken D'Ambrosio wrote:
To say that I'm confused would be understating things rather severely.
To say that we can't help you without seeing your config files would
also be an understatement. Unfortunately, we are not all-knowing nor
telepathic, so just saying it doesn't work won't generate much
To say that I'm confused would be understating things rather severely.
Kevin P. Fleming wrote:
To say that we can't help you without seeing your config files would
also be an understatement. Unfortunately, we are not all-knowing nor
telepathic, so just saying it doesn't work won't generate
Ken D'Ambrosio wrote:
Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0
sources that I downloaded from Digium, my Polycom 300 works just fine.
If I build with either various CVS builds, or the 1.0.6 sources from
Digium, I get No compatible codecs!.
WTF?
I'm using -the
Ken D'Ambrosio wrote:
Anyway, bottom line: all's well, now. Had to enter some extra stuff
into my sip.conf file, but the big clue was the fact that it even
-could- have been a config file problem, so many thanks.
Yes, sometimes what happens is that you are accidentally using
undocumented
On Tue, 01 Mar 2005 16:31:20 -0500, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
To say that I'm confused would be understating things rather severely.
Kevin P. Fleming wrote:
To say that we can't help you without seeing your config files would
also be an understatement. Unfortunately,
Original Post
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other
this may help you
http://billing.mutualphone.com/phpBB2/viewtopic.php?t=78postdays=0postorder=ascstart=15
On Tue, 18 Jan 2005 10:23:45 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Original Post
I have an Asterisk related problem with mutualphone.
I can
AM
Subject: Re: [Asterisk-Users] No compatible codecs
I've heard problems with the Grandstream G729 and the new digium G729
by MAC ID. Could be a compatibility issue with the implementations.
Did you ever use the Grandstream against asterisk with the old
Voiceage G729? I've heard that works
: [Asterisk-Users] No compatible codecs
I am using the G729 stack from Intel with *.
But as far as I know the Grandstream can just connect with PCMU and * will
transcode the audio into G729, right?
Because I know that iaxcomm and SJPhone for sure do not support G729 but I
can connect
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.
A working phone
Any suggestions about what I can change to make this work?
Yes, you should get a G729 license for your Asterisk.
Cheers!
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To
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, January 16, 2005 11:46 PM
Subject: Re: [Asterisk-Users] No compatible codecs
Any suggestions about what I can change to make this work?
Yes, you should get a G729 license
I've heard problems with the Grandstream G729 and the new digium G729
by MAC ID. Could be a compatibility issue with the implementations.
Did you ever use the Grandstream against asterisk with the old
Voiceage G729? I've heard that works just fine.
-- William
Hi,
I have a Cisco 5300 which I want to make a call THROUGH the
Asterisk PBX (security) to an IP phone which supports g729, and vice versa.
Both Cisco and the phone talk this codec if I do not force the call to go
through *
However if I say canreinvite=no in the sip.conf for either
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