[Asterisk-Users] Calls to Cisco PSTN gateway
Jeremy Jones
jjones at westcomllc.com
Thu Apr 15 06:11:48 MST 2004
Make sure you don't have "videosupport=yes" in sip.conf when using
as5300. I found mine doesn't like that much & got that codec error.
Jeremy
-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Radius
Sent: Thursday, April 15, 2004 2:37 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Calls to Cisco PSTN gateway
Hi all,
A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes,
made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line
with errors as follows:
-- Executing Dial("SIP/ata186-c1cf",
"SIP/29086988 at 110.100.231.2:5060|30|r") in new stack
-- Called 29086988 at 110.100.231.2:5060
<mailto:29086988 at 110.100.231.2:5060>
Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp: Error
in codec string 'ideo 0 '
Asterisk was configured with allow=ulaw. Any idea for this problem??
Thanks.
Ben
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