Re: Static SIP Service - Page 2

Yort Mantup · ‎02-05-2019

Moving from a dynamic SIP solution to static SIP. I completed the setup and have the sip trunk registered and when making a call to the test number assigned to the SIP trunk, my phone rings but the call does not pick up and eventually goes busy. My first thought is it's a codec issue but I have not found where that is the case.

ITSP --> SIP --> CUBE --> SIP --> CUCM --> IP Phone

I attached the debug for the incoming call.

Yort Mantup · ‎02-07-2019

I am expecting one of the 20x dial peers to get matched as well. So the incoming call should hit dial peer 100 first which connects the call to CUCM right? But there should be a second leg using a 20x dial peer back to the sip carrier right? I am not seeing that. I attached the output show only the debug voip ccapi inout

PierMatteo · ‎02-07-2019

So you match the CUCM dial-peer because of the destination pattern. However, you should actually match the 200 dial-peer based on the "incoming uri via" which is preferenced over destination-pattern. Can you check (sh dial-peer voice summary) if the 200 really is up and running? When calling inbound your matching should be: Mobile --> Provider --> Dial-Peer 200 --> Dial-Peer 100 --> CUCM. For responses to Provider in this scenario the sip-profile inbound on dial-peer 200 should be applied.

Yort Mantup · ‎02-08-2019

Good Call! Being that DP 200 is configured as below, I makes sense why it would not be up. I need to configure either an answer-address, incoming called number or destination pattern to the DP to go to an up status. Being that DP 200 is used to accept the call from the Provider as you stated, I should add the incoming called-number to capture everything. Correct?

dial-peer voice 200 voip
description Incoming dial-peer
voice-class sip profiles 201
voice-class sip profiles 200 inbound

PierMatteo · ‎02-08-2019

Yep- incoming called-number should do the trick and bring the dp up!

Yort Mantup · ‎02-08-2019

Ok, Using DP 200 now and not ringing my phone but instead getting the SIP/2.0 404 Not Found error in the log. I am fairly sure the CSS on the SIP trunk has access to the extension I am translating the DID to in CUCM. Still showing the 10.1.6.x IP addresses in the TO: and FROM: fields.

PierMatteo · ‎02-10-2019

Could you share debug ccsip messages and debug voip ccapi inout please?

Yort Mantup · ‎02-11-2019

see attached. thanks.

PierMatteo · ‎02-11-2019

The 404 is generated by CUBE not by CUCM. You can give this a try to fix it:

voice class dpg 100
dial-peer 100
!
dial-peer voice 200 voip
destination dpg 100
!

I think the SIP Profile works, as we can see in the TRYING message from CUBE, that From and To include the 209. IP address.

Yort Mantup · ‎02-11-2019

I believe your right about SIP profiles working. I made the change but am now getting 408 Request Timeout. Attached is new debug output. It looks like when using DP 100 to send the call out is causing the hang up. this is how DP100 is current configured. I have it bound to gig 1/0/1 which is the WAN interface.

dial-peer voice 100 voip
preference 4
session protocol sipv2
session target sip-server
incoming called-number 8T
voice-class codec 1
voice-class sip profiles 1 inbound
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-notify
fax-relay ecm disable
fax rate 14400
no vad

PierMatteo · ‎02-11-2019

Could you please share the current running-config with me?
I thought that DP 100 is pointing to CUCM.
Can you please change the dial-peer in the voice class dpg?
voice class dpg 100
no dial-peer 100
dial-peer <Enter-CUCM-DP>
!

Yort Mantup · ‎02-11-2019

Sorry about that.. I was making a few changes last week with DP 100. I set it back to where it was originally. Attached is current running config. The call should be using DP 100 to send the call out to the carrier correct?

Yort Mantup · ‎02-12-2019

Update. I am able to make outbound calls the new SIP Trunk. It is using DP 200. Inbound call to test number rings my phone but when I pick up there is no connection and my cell still shows calling.

combing thru the debugs I found that the SIP/2.0 Trying and Ringing are now showing the right IP address (209.x.x.x) but it is now the received portion which show the 10.1.x.x ip address which is the carriers internal IP's.

Is this still an issue on my end with the DP's or should I look to the carrier?

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 209.142.200.14:5060;branch=z9hG4bKdmrim7309otm0lighlo0.1

From: "" <sip:6162831487@209.142.200.14:5060>;tag=5679373

To: <sip:6162775094@209.142.200.14:5060>;tag=283044B8-1A57

Date: Tue, 12 Feb 2019 18:24:15 GMT

Call-ID: 1548058228-15951135@SFLDMIUP-C3SIPGW

CSeq: 1 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

P-Asserted-Identity: "" <sip:7771350@192.77.235.171>

Contact: <sip:6162775094@192.77.235.171:5060>

Server: Cisco-SIPGateway/IOS-16.6.4

Session-ID: 0000353e00105000a00080ce623bf878;remote=713a4e8f5fbc5c12a670ee27070804c2

Content-Length: 0

234795: Feb 12 2019 18:24:16.420 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:6162775094@192.77.235.171:5060 SIP/2.0

Via: SIP/2.0/UDP 209.142.200.14:5060;branch=z9hG4bKdmrim7309otm0lighlo0.1

To: <sip:6162775094@10.1.6.47:5060>

From: "" <sip:6162831487@10.1.6.4>;tag=5679373

Call-ID: 1548058228-15951135@SFLDMIUP-C3SIPGW

CSeq: 1 INVITE

Max-Forwards: 69

Contact: <sip:6162831487@209.142.200.14:5060;transport=udp>

Supported: 100rel

Expires: 330

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, PRACK, REFER, SUBSCRIBE, NOTIFY, UPDATE, REGISTER

Accept: application/sdp

Content-Type: application/sdp

Content-Length: 259

v=0

o=- 272265216 1549995856 IN IP4 209.142.200.14

PierMatteo · ‎02-07-2019

Is the provider's IP always the .14 or can it vary?

You could give this a go...:

-> profile 200 should copy the IP from the provider into a variable and you can use it to modify

-> profile 201 sets the .14 as a fixed IP

... maybe I'm missing something but I guess it' worth a try.

voice class sip-profiles 201
 rule 10 request ANY sip-header To modify "sip:(.*)@.*>" "sip:\1@209.142.200.14:5060>"
 rule 11 request ANY sip-header From modify "sip:(.*)@.*>" "sip:\1@209.142.200.14:5060>"
 rule 20 response ANY sip-header To modify "sip:(.*)@.*>" "sip:\1@209.142.200.14:5060>"
 rule 21 response ANY sip-header From modify "sip:(.*)@.*>" "sip:\1@209.142.200.14:5060>"
!
voice class sip-copylist 200
 sip-header from
 sip-header to
!
voice class sip-profiles 200
 rule 10 request INVITE peer-header sip To copy "sip:.*@(.*)>" u01
 rule 11 request INVITE peer-header sip From copy "sip:.*@(.*)>" u02
 rule 20 response ANY sip-header To modify "sip:(.*)@.*>" "sip:\1@\u01>"
 rule 21 response ANY sip-header From modify "sip:(.*)@.*>" "sip:\1@\u02>"
!
dial-peer voice 200 voip
 description Incoming dial-peer
 voice-class sip profiles 200 inbound
 voice-class sip profiles 201
!
dial-peer voice 201 voip
 description Outgoing 11 digit dial-peer
 voice-class sip profiles 201
!
dial-peer voice 202 voip
 description Outgoing 911 dial-peer
 voice-class sip profiles 201
!