[Asterisk-Users] Calls to Cisco PSTN gateway

Radius radius at broad-tel.com
Thu Apr 15 01:37:20 MST 2004


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
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040415/2da4d67e/attachment.htm


More information about the asterisk-users mailing list