configure sip in VG without cube to sip provider

[monasir]

Hi guys,

Can anyone help my out with this.
Recently a company has changed from SIP provider X to SIP provider Y
Voice traffic is  based on SIP without any cube.

SIP provider Y has provided me the signaling IP with port 5060
I have modified the dial-peers in that particular voicegateway, when i do a debug ccsip message i dont see any voice traffic.

In the dial-peers for incoming and outgoing calls i have changed the session target ip addres for the new SIP provider  please dial-peers below:
However i dont see any voicetraffic on the voice.
Did i forget something, what else can i troubleshoot??

dial-peer voice 10012 voip
 description **** company SIP provider outgoing international *****
 translation-profile outgoing VoIP-Interoute-Outgoing
 preference 1
 destination-pattern 00T
 session protocol sipv2
 session target ipv4:195.81.174.10
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
dial-peer voice 11010 voip
 description **** Company SIP provider incoming *****
 translation-profile incoming VoIP-Interoute-Incoming
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target ipv4:195.81.174.10
 incoming called-number 9604361[012].
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

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
 sip
  registrar server expires max 600 min 60
  no update-callerid

sip-ua
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 timers keepalive active 100
 sip-server ipv4:195.81.174.10:5060
connection-reuse
  host-registrar

They are not registrar so no username or password is required.

XXXXXX ( VG#195.81.174.10 5060
Trying 195.81.174.10, 5060 ... Open

Any help would be appreciated.

[Manish Gogna]

Hi,

Can you do a shut / no shut on the dial-peers and check if any output is generated for "debug ccsip messages" for a test call?

Manish

[Chris Deren]

Slightly confused here as you claim there is no CUBE, yet you provided CUBE configuration. Please provide more details on your topology.

Also, provide the output of the "debug ccsip messages" for outbound so we can at least see if there is anything going out, please also provide "debug voice ccapi inout"

[Ryan Huff]

I would step through each item here, placing a test call after each (with debug ccsip messages turned on) and see if you are able to see SIP messages.

• Double check the IP address, verify that 195.81.174.10:5060 is correct
• Increase the logging buffer (especially if you have high traffic on the router)
• conf t
• logging buffer 10000000 (example)
• Shut/No shut the dial-peers
• If the platform is at code release 15.1(2)T or later add:
• conf t
• voice service voip
• ip address trusted list
• ipv4 195.81.174.10 255.255.255.255
• Although not having the ITSP peer address(es) in the ip address trust list wouldn't stop SIP messaging from showing in the logs (and would give you much different problems), it is worth verifying that you have it added if the platform supports it.
• Remove the user agent (sip-ua) configuration then add it back
• Schedule a reload of the router