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ITSP sending username, not called number. inbound sip-profile issues

etechexperts
Level 1
Level 1

I'm setting up SIP trunks on a CME system using sip phones. I've updated the IOS to c2900-universalk9-mz.SPA.157-3.M8.bin to take advantage of SIP tenant profiles. The SIP trunk appears to be up and registered.

CME#show sip-ua register status tenant 2000

Tenant: 2000
--------------------- Registrar-Index 1 ---------------------
Line                                                           peer            expires(sec)      reg  survival      P-Associ-URI
================================ ========= ============ === ======== ============
ITSPusername                                            -1               83                     yes  normal

The SIP provider has provided a test phone number. When I call the number I receive a fast busy signal. Debug ccsip messages is showing that the provider is sending the username in the INVIITE Req-URI field and a debug voice dialpeer shows nothing matching as a result. I have talked to the provider and they basically said that's how they send INVITES to all of their customers and that the TO field is correct in their header. They won't change their headers and I need to adjust my SIP headers accordingly.


There is a Cisco ASA in front of the Cisco 2900 CME router. The provider is listening on udp port 5065. The CME is using 5060. I've created SIP profiles based on online documentation, but I'm not having any luck. It doesn't look like the inbound SIP profile command is picking up the call. I'm attaching a config and a few debugs so that maybe a few new sets of eyes can see what is going on. I appreciate any assistance. thank you.

1 Accepted Solution

Accepted Solutions

b.winter
VIP
VIP

You have to activate the "SIP profile inbound" feature first.

voice service voip
 sip
  sip-profiles inbound

Furthermore, it would be better to match the inbound call from the ITSP based on the IP / FQDN of the ITSP, and not based on any number.

voice class uri 15 sip
 host ipv4:<IP>
 or
 host dns:<fqdn>
!
dial-peer voice 15 voip
 incoming uri via 15

View solution in original post

4 Replies 4

b.winter
VIP
VIP

You have to activate the "SIP profile inbound" feature first.

voice service voip
 sip
  sip-profiles inbound

Furthermore, it would be better to match the inbound call from the ITSP based on the IP / FQDN of the ITSP, and not based on any number.

voice class uri 15 sip
 host ipv4:<IP>
 or
 host dns:<fqdn>
!
dial-peer voice 15 voip
 incoming uri via 15

etechexperts
Level 1
Level 1

Thanks for the help. I was able to get this working.

I didn't realize I had to "activate" the inbound feature.

I added the sip inbound feature to voice service voip and then added voice-class sip profiles 100 inbound to my incoming SIP dial peer. I also added the voice class uri based on the IP and it started working.

Working config:

voice service voip
sip
sip-profiles inbound

voice class uri 15 sip
host ipv4:<<ITSP Public IP>>

voice class sip-profiles 100
request INVITE sip-header To copy "sip:(.*)@" u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"

dial-peer voice 15 voip
description SIP Trunk - Incoming
translation-profile incoming SIPTRUNKIN
session protocol sipv2
session target sip-server
session transport udp
incoming uri via 15
voice-class codec 1
voice-class sip profiles 100 inbound
voice-class sip tenant 2000
voice-class sip bind control source-interface GigabitEthernet0/1.1
voice-class sip bind media source-interface GigabitEthernet0/1.1
dtmf-relay rtp-nte
no vad

Glad you got it to work! One comment on your shared configuration. On your inbound dial peer, DP 15, there is no need for this line “session target sip-server” as that is used on outbound dial peers.



Response Signature


If your problem could be resolved, then I would really appreciate an "accepted solution".