Hi Ryan, Do you see a codec specified in SDP content of the INVITE sent to CM when calling from FXS to CM?
Rob On Mon, Dec 15, 2008 at 12:45 AM, Ryan Trauernicht <ryanstudyvo...@gmail.com > wrote: > SIP analog phone that hangs off an FXS port. I created a SIP trunk (MTP > required) in CM. It is set to G711ulaw only region. Cisco IP Phones can > call the analog phone just fine, but the analog phone can not call the IP > Phones. > > CSS is applied to the Sip trunk and significant digits are set to 4. > > voice service voip > allow-connections h323 to sip > allow-connections sip to h323 > allow-connections sip to sip > h323 > > > > dial-peer voice 100 pots > destination-pattern 2100 > progress_ind setup enable 3 > port 2/0 > ! > dial-peer voice 101 voip > destination-pattern [23]...$ > session protocol sipv2 > session target ipv4:192.168.187.220 > incoming called-number 2100 > dtmf-relay sip-notify > codec g711ulaw > ip qos dscp cs3 signaling > no vad > > The thing that sticks out to me is the "debug ccsip all" shows that it is > trying to negotiate at no codec: > > > Dec 15 00:40:50.926: //16/D965F838801A/SIP/Call/sipSPIMediaCallInfo: > Number of Media Streams: 1 > Media Stream : 1 > Negotiated Codec : No Codec > Negotiated Codec Bytes : 0 > Negotiated Dtmf-relay : 0 > Dtmf-relay Payload : 0 > Source IP Address (Media): 10.0.0.2 > Source IP Port (Media): 18568 > Destn IP Address (Media): 0.0.0.0 > Destn IP Port (Media): 0 > Orig Destn IP Address:Port (Media): 0.0.0.0:0 > > > > Any ideas? > > Thanks, > Ryan Trauernicht >
