El vie, nov 17 de 2006 a las 04:34 +0530, Vicky comentaba:
Thats really strange .. if you have made canreinvite=no then it should not
even attampt native bridging and should transcode codecs ..something's fishy
here .. Also try to put canreinvite=no in testulaw exntension too .
So why do I
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont edit/transcode
stream .
On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote:
I have
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then asterisk will just do bridging of g729 and wont edit/transcode
stream .
El jue, nov 16 de 2006 a las 11:35 -0600, Victor Toofic comentaba:
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per
channel license . If you are just using asterisk and havent bought g729
license then
Thats really strange .. if you have made canreinvite=no then it should not
even attampt native bridging and should transcode codecs ..something's fishy
here .. Also try to put canreinvite=no in testulaw exntension too .
On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote:
El jue, nov 16 de 2006
I have the following scenario:
g729gsm
UAS --- * --- UAC
I am using sipp to generate the calls between the UAC and the UAS and
sending some rtp from the UAC, I want * to do transcoding but as I see
it is not. As long as I know 'Attempting native bridge'
Hello list,
I am encountering a bit of a problem in working with incoming calls with a
TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the
ringing, but will sometimes report multiple Attempting native bridge.
What I do is basically that when a call comes in, I dial a
Hi,
I have problems with two trunks, ZAP3 and
ZAP4. ZAP4 is connected to PSTN line while ZAP3 is connected to analogical
switchboard.
The
system is able to redirect calls from ZAP4 to ZAP3, through an
IVR, but, hanging up doesnt work .
This is the CLI report where you can see, at
On Apr 12, 2005, at 9:38 PM, snacktime wrote:
That would be great if I didn't want * to get out of the media path,
but I do. In my case everything works great with the teliax 800 DID,
but not with the local number DID. I think it's an issue on their end
myself.
I didn't want to insinuate that Teliax was in any way sloppy, but they
*are* the ITSP I was referring to when I mentioned earlier in this
thread that my provider was having issues with native bridging.
I raised a ticket with them and they're working on resolving the bug
currently, so I
Hello
We find an issue when IAX wants to transfer the native bridge. We are using
asterisk 1.0.7.
Asterisk shows following messages after getting 'answered'.
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and
IAX2/xxx.xxx.xxx.xxx:/3
-- Channel 'IAX2/[EMAIL PROTECTED]/2' ready
On Apr 12, 2005, at 11:21 AM, Xu Wang wrote:
Hello
We find an issue when IAX wants to transfer the native bridge. We are
using
asterisk 1.0.7.
Asterisk shows following messages after getting 'answered'.
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and
IAX2/xxx.xxx.xxx.xxx:/3
. But the
call can still continue.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Goodyear
Sent: Tuesday, April 12, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Attempting native bridge
On Apr 12, 2005, at 1:35 PM, Xu Wang wrote:
i do have 'answer' to 1st incoming IAX before calling the 2nd IAX.
Yes, the
log looks almost the same. I have 1800 coming from one vendor, then
call
through 2nd vendor (it might be the same vendor as the 1st ) to the
destination.
If 'attempting native
-Users] Attempting native bridge of
On Apr 12, 2005, at 1:35 PM, Xu Wang wrote:
i do have 'answer' to 1st incoming IAX before calling the 2nd IAX.
Yes, the
log looks almost the same. I have 1800 coming from one vendor, then
call
through 2nd vendor (it might be the same vendor as the 1st
I also have a native bridge problem. I have 2 analogue phones each
connected to an IAXy. When attempting a call between them I get the
following:
-- Accepting DIAL from nnn.nnn.117.75, formats = 0x4
-- Executing Dial(IAX2/[EMAIL PROTECTED]/6, IAX2/kitchen) in
new stack
-- Called
On Apr 12, 2005, at 3:31 PM, Mike Price wrote:
I also have a native bridge problem. I have 2 analogue phones each
connected to an IAXy. When attempting a call between them I get the
following:
-- Accepting DIAL from nnn.nnn.117.75, formats = 0x4
-- Executing Dial(IAX2/[EMAIL PROTECTED]/6,
I have a very strange bridging problem also with teliax. I have an
800 DID and a local number DID with them. Both numbers go to the same
context, where the caller is dropped into DISA, and the outgoing call
also goes out through teliax.
When dialing into the 800 number, everything works. When
] Attempting native bridge of
I have a very strange bridging problem also with teliax. I have an
800 DID and a local number DID with them. Both numbers go to the same
context, where the caller is dropped into DISA, and the outgoing call
also goes out through teliax.
When dialing into the 800 number
On 4/12/05, Xu Wang [EMAIL PROTECTED] wrote:
add following line in the context of IAX.conf
NOTRANSFER=YES
That would be great if I didn't want * to get out of the media path,
but I do. In my case everything works great with the teliax 800 DID,
but not with the local number DID. I think
ERROR CONDITION
---
-- Executing Dial(SIP/2001-f6c4, SIP/2000|20) in new stack
-- Called 2000
-- SIP/2000-0ead is ringing
-- SIP/2000-0ead answered SIP/2001-f6c4
-- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead
Have searched web and archive w/o good
Hi,
Hope somebody has an idea as to what the following means:
I am making a call from one xlite client (2000) to another xlite
client (2001) via asterisk. The call seems to connect fine and each
client comes up as 'connected'. They both have the same codecs
enabled and have turned the silence
The audio is carried on two RTP streams: one for each direction. Is it
possible those streams are being blocked by a firewall or something of
the sort?
The attempting native bridge message means that Asterisk is bridging
the two calls together without doing any codec translation... uLaw to
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