asterisk users - Jul 2006 - 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

Hi Luca, 
are you using SIP reinvite ?

post a bit mor information (sip.conf)


Fabio

-----Mensaje original-----
De: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]En 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

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users