Outbound calls not going on SIP trunk with CME

[Alexander Nejo]

Hello Everyone, i have a sip trunk on CME, and inbound calls was going initially, but the outbound was having issue. The SIP trunk is registered, but i can see "sip/2.0 500 internal server error" on the debugs

I have limited knowlegde on cme. 

Kindly advice on what i can do to resolve the outbound and inbound calls.

Attached are the configs and ccsip debugs

[b.winter]

Hi,
please share the config and the debug output. There is nothing attached to your original post.

[Alexander Nejo]

Apologies, pls is the config and the Debugs

[Alexander Nejo]

The Debug

[b.winter]

Is this a new installation or was this already working?
Do you have any information from the provider (e.g. how an outgoing invite has to look like?)

Another thing:
When having a SIP trunk and CME co-resistant on one router, you need to work with tenants instead of the sip trunk details in the SIP-UA section: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/ios-xe/config/ios-xe-book/m_voi-cube-multi-tenants.html

What version does your router have?

Config could look like this:

voice class tenant 1000
 registrar 1 dns:sip.ipnxtelecoms.com expires 300
 credentials username 016321586 password 7 114A4B5D25231E0657107A332760 realm ipnxnigeria.net
 authentication username 016321586 password 7 0245560339371A2B1F74580E0A44 realm ipnxnigeria.net
 session refresh
 error-passthru
 sip-server dns:sip.ipnxtelecoms.com:5060
 outbound-proxy dns:sip.ipnxtelecoms.com:5060
 no remote-party-id
 bind control source-interface GigabitEthernet0/0/0.10
 bind media source-interface GigabitEthernet0/0/0.10
 no pass-thru content custom-sdp
 privacy-policy passthru
!
dial-peer voice 100 voip
 translation-profile incoming pstn-incoming-XXXX
 session protocol sipv2
 ### session target sip-server --> not necessary on an incoming dial-peer
 incoming called-number .
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 no vad
 ### authentication username 016321586 password 7 100D5B41372607015F3E7B3C2B7B realm ipnxnigeria.net --> already in the tenant
 ### voice-class sip tenant 1000
!
dial-peer voice 10 voip
 description LANDLINE
 destination-pattern 016321586
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 no vad
 ### authentication username 016321586 password 7 100D5B41372607015F3E7B3C2B7B realm ipnxnigeria.net --> already in the tenant
 ### voice-class sip tenant 1000
!
dial-peer voice 20 voip
 description ***OUTBOUND CALS***
 translation-profile outgoing pstn-outgoing-XXXX
 destination-pattern ...........
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 no vad
 ### authentication username 016321586 password 7 100D5B41372607015F3E7B3C2B7B realm ipnxnigeria.net --> already in the tenant
 ### voice-class sip tenant 1000
!
### why you have 2 incoming dial-peers?
### why is there a different realm name?
dial-peer voice 2 voip
 description ***INBOUND CALLS***
 translation-profile incoming IN
 session protocol sipv2
 ### session target sip-server --> not necessary on an incoming dial-peer
 incoming called-number .
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte
 no vad
 authentication username 016321586 password 7 0047415E366A1E0C5C1B1D59064A realm sip.ipnxtelecoms.com
!
!
sip-ua
 no remote-party-id
 retry invite 2
 retry register 2
 retry options 1
 timers connect 100
 host-registrar
 presence enable

[Alexander Nejo]

Thanks for your swift response. The version my router (4331) is using is Version 15.5(3)S4b, and its not taking "Voice class tenant command

[b.winter]

It's not taking the command, because the version is veeeery old.

I would recommend to upgrade it to at least this version: https://software.cisco.com/download/home/285018115/type/282046477/release/Fuji-16.9.8
This is the last version without smart licensing.

[Alexander Nejo]

Hi Advocate,

Thanks for your response,. The update looks like a long route to take,. Could we have a work around or something else?

[b.winter]

I don't know, why it should be a long route. Update + reconfiguration about half hour.

But about your debug:
Who is sending the initial invite? You say, outbound calls are not working, but the INVITE I see, seems to be an incoming INVITE from external. Is this the correct outgoing call you traced?

Moreover, the lines always only start with 1 letter:

INVITE sip:016321586@10.100.10.1:5060 SIP/2.0
v: SIP/2.0/UDP 41.184.56.164:5060;branch=z9hG4bK242162;rport
Record-Route: <sip:41.184.56.164:5060;lr>
Record-Route: <sip:rev.922835.dialog.cgatepro;lr>
v: SIP/2.0/UDP 62.173.33.254:5060;branch=z9hG4bK59422-aoklssp;cgp=ipnxnigeria.net;rport=5060
Record-Route: <sip:62.173.33.254:5060;lr>,<sip:rev.9417881-10.51.2.3.dialog.cgatepro;lr>
Max-Forwards: 66
f: "07037858093" <sip:07037858093@ipnxng.net>;tag=82635D36-113526-9411CAAB_aoklssp-1724
t: <sip:16321586@telnum>
i: 1e2afa73-1a5.gwin@ipNXSs
m: <sip:signode-113526-9411CAAB_aoklssp-1724@62.173.33.254>
CSeq: 1 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
User-Agent: CommuniGatePro-callLeg/5.4.6
x: 3600
Min-SE: 180
Content-Type: application/sdp
l: 585 

What is your complete setup? Is your CME behind a FW, with NAT to the public IP?
If yes, does your FW do SIP ALG?

[Alexander Nejo]

There is no firewall and the setup is with NAT to public IP.

Attached is the debug for outbound calls,. Incoming calls are working fine now

[Roger Kallberg]

As your version is past everything the recommendation would be to upgrade it to something more current. We use version 17.6.4 on our ISR4K voice gateways and it works well.

[Alexander Nejo]

Dear Roger,

I have done the upgrade and added the voice class tenant,.. The sip trunk is registered and the outbound is fine the inbound calls are ringing and no audio. 

Attached is the latest config and the debug.

[Roger Kallberg]

Before looking at your shared configuration and debugs I have a question. Initially you said that inbound calls were working, but you had issues with outbound. Now if I’m not reading your message incorrectly the situation is the reverse. If so can you please outline exactly what changes you have done to the configuration so that we don’t need to chase them down by comparing what you initially shared with your current post?

[Roger Kallberg]

This article might be helpful for you to track down the cause(s) of no audio. https://community.cisco.com/t5/collaboration-knowledge-base/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442/jump-to/first-unread-message

As you mentioned that you use NAT you’ll need to use a SIP profile for changing the content of the SDP in the SIP dialogue as NAT only changes the IP information in the outer layer of fields and doesn’t touch any information in the actual payload of the communication. This causes the endpoints to try to communicate with the original IP addresses, ie pre NAT IPs and that doesn’t work. If you search the community you’ll find plenty of post on this topic as it’s a pretty common thing to ask.