Bind SIP control and media to the IP defined in the SIP trunk:

Voice service voip
 sip
  bind all....

Ensue that the FXS port is not registered to GK:

Dial-peer voice xx port
 no e164 reg

Ensure that the MTP is correctly configured (in a Device Pool which
communicates using g711u).


-- 
Vik Malhi ­ CCIE #13890, CCSI #31584
Senior Technical Instructor - IPexpert, Inc.

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From: Ryan Trauernicht <ryanstudyvo...@gmail.com>
Date: Sun, 14 Dec 2008 18:45:49 -0600
To: OSL CCIE Voice Lab Exam <ccie_voice@onlinestudylist.com>
Subject: [OSL | CCIE_Voice]  SIP Analog phone fast busy

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 <http://0.0.0.0:0>
 
 
 
Any ideas?
 
Thanks,
Ryan Trauernicht


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