[Asterisk-Users] codec negotiation

[Eduardo Goncalves]

Hi list,

I'm with a little problem on codec negotiation between a cisco827 and
asterisk.

My sip.conf is like that:

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all

and cisco like that:

dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
dtmf-relay rtp-nte
codec g711alaw
no vad
!

When I try to make a call, cisco shows codec g711alaw, but asterisk
shows codec g729A (i have the licenses) and there is no audio. When I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.

The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok).

Why this negotiation problem happens?

Thanks
Eduardo

[Hector Q.-datafull]

I'm working with DIA096B on two remote computers that are behind NAT. They register ok.
The * has a static public IP address.
I saw other simliar posts btu this seems to be different.

The call is from test2 --> test3:

-- Accepting AUTHENTICATED call from xx.xx.xx.xx , requested format = 2, actual format = 2
-- Executing Dial("IAX2[***@test3]/3", "IAX/test2") in new stack
NOTICE[1200884528]: File app_dial.c, Line 506 (dial_exec): Unable to create channel of type 'IAX'
== Everyone is busy at this time
-- Executing VoiceMail("IAX2[***@test3]/3", "u2002") in new stack
xx.xx.xx.xx is the public ip of the NAT in front of DIAX096B (test2)

[general]
port=5036
bindaddr=publicipaddress
disallow=all
allow=gsm
jitterbuffer=3
tos=reliability

[test2]
type=friend
username=test2
secret=........
host=dynamic
context=test

[tito3]
type=friend
username=test3
secret=......
host=dynamic
context=test


Thanks.
HQ.

[Andrew Thompson]

----- Original Message -----
From: "Eduardo Goncalves" <***@acenet.com.br>
To: <asterisk-***@lists.digium.com>
Sent: Tuesday, December 16, 2003 1:08 PM
Subject: [Asterisk-Users] codec negotiation

Post by Eduardo Goncalves
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all
dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
dtmf-relay rtp-nte
codec g711alaw
no vad
!
When I try to make a call, cisco shows codec g711alaw, but asterisk
shows codec g729A (i have the licenses) and there is no audio. When I
try disallow=g729, the same occurs, but this time asterisk shows codec
gsm.
The only way to make a call is allowing only alaw. But this is not
convenience, since i need to use g279 with another endpoint (working
ok).

You could try setting the codec before dialing that particular provider.
Except I don't see the command now that I'm trying to find it...

Post by Eduardo Goncalves
Why this negotiation problem happens?

Can't help on that one...

Post by Eduardo Goncalves
Thanks
Eduardo

-----
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.

[Nguyen Hoang Lan]

Hello Eduardo,

Wednesday, December 17, 2003, 1:08:00 AM, you wrote:

EG> Hi list,

EG> I'm with a little problem on codec negotiation between a cisco827 and
EG> asterisk.

EG> My sip.conf is like that:

EG> [general]
EG> port = 5060
EG> bindaddr = 0.0.0.0
EG> context = default
EG> amaflags = default
EG> allow=g729
EG> allow=gsm
EG> allow=alaw
EG> allow=ulaw
EG> ;disallow=all

EG> and cisco like that:

EG> dial-peer voice 6 voip
EG> destination-pattern 0T
EG> session protocol sipv2
EG> session target ipv4:<asterisk-ip>
EG> dtmf-relay rtp-nte
EG> codec g711alaw
EG> no vad
EG> !

EG> When I try to make a call, cisco shows codec g711alaw, but asterisk
EG> shows codec g729A (i have the licenses) and there is no audio. When I
EG> try disallow=g729, the same occurs, but this time asterisk shows codec
EG> gsm.

EG> The only way to make a call is allowing only alaw. But this is not
EG> convenience, since i need to use g279 with another endpoint (working
EG> ok).

EG> Why this negotiation problem happens?

Try to add to cisco peer (not shown in your mail)

[cisco]
....
disallow=all
allow=alaw

[Jonathan Tew]

We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in "sip show
peers" with the IP address of their firewall, etc. They can receive
calls no problem. After some time goes by... they don't show as
registered with * any more in the sip show peers. They can still make
outbound calls, but can not receive the inbound ones. Anyone have any
ideas on this one?

Thanks,
Jonathan

[Eric Wieling]

Their firewall may be timeing them out. Try adding qualify=60 to each
of the entries in sip.conf

Post by Jonathan Tew
We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in "sip show
peers" with the IP address of their firewall, etc. They can receive
calls no problem. After some time goes by... they don't show as
registered with * any more in the sip show peers. They can still make
outbound calls, but can not receive the inbound ones. Anyone have any
ideas on this one?
Thanks,
Jonathan
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Jonathan Tew]

I think we've having some luck with this setting. Of course we had to
crank it up higher so that it didn't consider the clients LAGGED. When
the clients were LAGGED they couldn't receive any calls for some
reason. So like a setting of 200ms seems to work fine for everyone.

Post by Eric Wieling
Their firewall may be timeing them out. Try adding qualify=60 to each
of the entries in sip.conf

[Jonathan Tew]

At my home office I have a X100P card in a server that I've been using
for testing. The machine it is in is connected to a HP fax machine and
then to the wall outlet. This morning the SBC installer showed up at my
house for the ADSL install on that line. He said they detected a
short. So he tested the outside box and it was fine. He said it was
inside. So we came inside and tested the two devices with his little
box. The fax was fine... the X100P card however was causing the short.
Now of course I'm going to install a filter on this line for the ADSL,
but is this short normal? The installer says that it will kill the ADSL
signal. Maybe the X100P is defective? It's been working fine though
for making and answering calls to this point.

Thanks,
Jonathan

[Martin Pycko]

The registry expires after sime time. You can set the default expirey and
max in sip.conf. It's up to your phone/sip device to reregister after the
registration expires.

Martin

Post by Jonathan Tew
We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in "sip show
peers" with the IP address of their firewall, etc. They can receive
calls no problem. After some time goes by... they don't show as
registered with * any more in the sip show peers. They can still make
outbound calls, but can not receive the inbound ones. Anyone have any
ideas on this one?
Thanks,
Jonathan
_______________________________________________
Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

[Chris Albertson]

My Strandstream BT100 is working OK for both inbound and outbound now
except that when you speak into the handset you cannot hear your
own voice in the earpeice. It works OK, the other end can hear the
call but most telephone users have become used to hearing their own
voice.

Is this something I can fix or is it a "feature" of the GS phone?




=====
Chris Albertson
Home: 310-376-1029 ***@yahoo.com
Cell: 310-990-7550
Office: 310-336-5189 ***@aero.org
KG6OMK

__________________________________
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New Yahoo! Photos - easier uploading and sharing.
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[Stephen R. Besch]

Post by Chris Albertson
My Strandstream BT100 is working OK for both inbound and outbound now
except that when you speak into the handset you cannot hear your
own voice in the earpeice. It works OK, the other end can hear the
call but most telephone users have become used to hearing their own
voice.
Is this something I can fix or is it a "feature" of the GS phone?

"Sidetone" (hearing yourself talk) should be generated by the phone. I
suspect that if it is not there, then the phone is defective and should
be replaced. It is not something that is configurable, as far as I
know. I have 20 of them on site and they all generate proper sidetone
out of the box. The only other possibility I can think of is that
silence supression (a GS option) is enabled AND also not working
properly. I have never experimented with silence supression. Anyone
wand to pipe in?

Stephen R. Besch

[Alfred R. Nurnberger]

My guess would be that the NAT firewall times out and closes the port.
Reopening it from the inside is no problem, but access from the outside gets
blocked.
In order to keep the path open both ways, the client needs to send some kind
of messages with the proper IP/port in regular intervals.

Alfred.

Post by Jonathan Tew
We have people connecting to an asterisk box over the internet. They're
using the x-lite client behind linksys firewalls. The X-Lite client
discovers the firewall no problem and connects to Asterisk without a
problem. After connecting the agent shows up properly in "sip show
peers" with the IP address of their firewall, etc. They can receive
calls no problem. After some time goes by... they don't show as
registered with * any more in the sip show peers. They can still make
outbound calls, but can not receive the inbound ones. Anyone have any
ideas on this one?
Thanks,
Jonathan

[Johannes von Drachenfels]

Hi,

i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
need is to send out the extension of the inside callers to, for example:
78107-14

So what i tried is:

exten => _00XX.,1,SetCallerID(78107${CALLERIDNUM})
exten => _00XX.,2,Dial,Zap/g1/${EXTEN:2}

And this is what my asterisk ist telling me:

-- Executing SetCallerID("SIP/14-9707", "7810714") in new stack
-- Executing Dial("SIP/14-9707", "Zap/g1/1716710815") in new stack
-- Called g1/1716710815
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'

But i still can see only the 78107-0 when i call my mobile ...

Any help would be great ...

Thanks, Johannes

[Peer Oliver schmidt]

Post by Johannes von Drachenfels
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
78107-14

[..]

Post by Johannes von Drachenfels
But i still can see only the 78107-0 when i call my mobile ...

Might this be a configuration issue on the Telekom side of things? Pure
speculation on my part.

rgds
pos

[Johannes von Drachenfels]

sorry, after some time and a couple of beers i found the solution by myself
...
In germany we have to set the callerid in national style like 72317810714
cause otherwise the telekom will change it again ...

Thanks for help,

Johannes

-----Ursprungliche Nachricht-----
von Drachenfels
Gesendet: Montag, 9. Februar 2004 18:33
Betreff: [Asterisk-Users] no extension in callerid of outgoing calls ...
Hi,
i'm here in germany still fighting against my problems ...
We have a e100p which is sending out his callerid as 78107-0. But what i
78107-14
exten => _00XX.,1,SetCallerID(78107${CALLERIDNUM})
exten => _00XX.,2,Dial,Zap/g1/${EXTEN:2}
-- Executing SetCallerID("SIP/14-9707", "7810714") in new stack
-- Executing Dial("SIP/14-9707", "Zap/g1/1716710815") in new stack
-- Called g1/1716710815
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
But i still can see only the 78107-0 when i call my mobile ...
Any help would be great ...
Thanks, Johannes
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[dkwok]

I have outgoing connection to iaxtel and another iax server A.

iax server A only accept g729 codec while iaxtel is something I am not
quite sure of. At the moment iaxtel only accepts gsm. I remember
previously it does accept g729.

my problem due to the switching between codec when making outgoing calls
to these servers.

my iax.conf has these lines:

[general]

disallow=all
allow=gsm
allow=g729

I believe the general context define the codec to be used when making
outgoing calls. The peer context below general context is to governed
codec to be used for incoming calls. Is this correct?

now if I specificly disallow g729 in the general context I can make
calls to iaxtel. however i cannot make calls to server A as it only
accepts g729. After I allow g729, I can make call to server A but the
call made to iaxtel cannot go through.

The console indicates that the call is accepted by iaxtel using codec
729A, then it says the circuit is too busy.

Is there a clever way of governing the codec use for each outgoing
connection in order to avoid the issue in codec negotiation?

[Michael Graves]

Why do you need 729? I just called your IAXTel number using GSM and
connected fine.

Michael

Post by dkwok
I have outgoing connection to iaxtel and another iax server A.
iax server A only accept g729 codec while iaxtel is something I am not
quite sure of. At the moment iaxtel only accepts gsm. I remember
previously it does accept g729.
my problem due to the switching between codec when making outgoing calls
to these servers.
[general]
disallow=all
allow=gsm
allow=g729
I believe the general context define the codec to be used when making
outgoing calls. The peer context below general context is to governed
codec to be used for incoming calls. Is this correct?
now if I specificly disallow g729 in the general context I can make
calls to iaxtel. however i cannot make calls to server A as it only
accepts g729. After I allow g729, I can make call to server A but the
call made to iaxtel cannot go through.
The console indicates that the call is accepted by iaxtel using codec
729A, then it says the circuit is too busy.
Is there a clever way of governing the codec use for each outgoing
connection in order to avoid the issue in codec negotiation?
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. ***@mstvp.com

"I believe there comes a time when everything just falls in line."
- Pat Benatar, from All Fired Up.

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