RE: [asterisk-users] audio session start delay

2006-07-09 Thread Luca Corti 
On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote:
 are you using SIP reinvite ?

Proably not as I'm using t in Dial()s for call transfer.

 post a bit more information (sip.conf)

[general]
context=sip
allowguest=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
domain=mydomain.com
domain=1.2.3.4
allowexternalinvites=no
language=it
relaxdtmf=yes

[authentication]

[as5350] ; My PSTN gateway
type=peer
qualify=200
host=1.2.3.5
fromdomain=1.2.3.5
insecure=very

[ser] ; My SIP proxy
type=peer
qualify=200
host=1.2.3.6
fromdomain=1.2.3.6
insecure=very

[01]; Extension example
callerid=My Name 01
nat=yes
type=friend
username=01
secret=mypass
host=dynamic
dtmfmode=rfc2833
context=uffici
canreinvite=no
callgroup=1
pickupgroup=1
qualify=no

Thanks

-- 
Luca Corti
PGP Key ID 1F38C091
Adesso dico: L'usignolo chiuso in gabbia smette di cantare.

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RE: [asterisk-users] audio session start delay

2006-07-06 Thread Fabio 
Hi Luca, 
are you using SIP reinvite ?

post a bit mor information (sip.conf)


Fabio

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Luca Corti
Enviado el: Jueves, 06 de Julio de 2006 01:59 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] audio session start delay


Hello everyone,

I've set up an asterisk box with basic PBX features (DiD, MoH, MoT,
Blind and Attended Call Transfer, PickUp, ecc.) for 10 Cisco 79xx (7912
and 7960) with SIP image (8.0). PSTN gateway is done using a Cisco
AS5350 with two ISDN PRIs connected to Asterisk via SIP. Between the
phones and the PBX I have a router doing NAT and a 4mbit synchronous
line.

Sometimes when calling between extensions, after successful signaling,
there is a delay of 10 seconds before any audio is heard by both
parties.

Do you know what can cause this behaviour? Is this more likely to be a
phone or an Asterisk issue?

thanks

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