Hello,
I am trying to get a SIP trunk working but am unable to make or recieve calls. Our setup is:
SIP Trunk (Provider) ---> SIP Gateway (Me) -----> Cisco Gateway (Me) --- CUCM
The SIP gateway is a cisco 2900 device and is completely separate from our network. Its job is to send and recieve calls on the SIP trunk. There is a ISDN QSIG Pri line configured from this SIP gateway to another of our voice gateways then to deliver calls to the IP phones via MGCP.
Anyway, the SIP trunk is not working. Below is a copy of my config (some of it). I also attached a copy of the calls logs. Can someone help me interpret the SIP call logs to try to narrow down why the call fails?
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server
asserted-id pai
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
<snip>
!!
controller E1 0/1/0
framing NO-CRC4
pri-group timeslots 1-31
description :: E1 PRI link to VOICE GW
!
dial-peer voice 1000 voip
translation-profile outgoing OutBound_CallerID_SIP
destination-pattern .T
session protocol sipv2
session target ipv4:10.xx.xx.xxx:5060
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 pots
destination-pattern 358.........
incoming called-number .
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 101 pots
description **Dial-Peer to match Incoming calls from SIP TRUNK**
destination-pattern 12345...
incoming called-number .
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 102 pots
description **Dial-Peer to match Incoming calls from SIP TRUNK**
destination-pattern 35812345...
incoming called-number .
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 1001 voip
translation-profile outgoing OutBound_CallerID_SIP
description "TEST DIAL PEER"
destination-pattern 0044.........
session protocol sipv2
session target ipv4:10.xx.xx.xxx:5060
codec g711alaw
!
!
sip-ua
Here is the CCSIP call log failure:
The Call Setup Information is:
Call Control Block (CCB) : 0x40ADE120
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +358xxxxxxxxx
Called Number : 003xxxxxxxxxxx
Source IP Address (Sig ): 10.xx.xx.xx
Destn SIP Req Addr:Port : 10.xx.xx.214:5060
Destn SIP Resp Addr:Port : 10.xx.xx.214:5060
Destination Name : 10.xx.xx.214
*Dec 3 09:22:44 UTC: //12659/3E5136E58059/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.xx.xx.xx
Source IP Port (Media): 17650
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Dec 3 09:22:44 UTC: //12659/3E5136E58059/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 127
Disconnect Cause (SIP) : 400
