Outbound Call SIP

[Leonardo Santana]

Hello,

We configured a integration with the PSTN using a SIP Trunk.

Call Flow:

IP Phone -> CUCM -> SIP -> Voice Gateway -> SIP Trunk -> PSTN

On the debug ccsip messages i see that our side is sending a cancel message to the ITSP.

Why we are sending this?

The debug and config are attached.

Thanks for the help

Leonardo Santana

[Nadeem Ahmed · Accepted solution]

Do you mind collecting another trace with following debugs

debug voice ccapi inout

debug ip tcp transcation

debug ccsip message

One thing which i noticed where CUBE is not responding to INVITE received from CUCM. also on incoming leg you have TCP connection on outgoing towards ITSP you have UDP?

*Jan 11 23:26:06.495: //307/8AF73A800007/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:2430600@10.115.182.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.115.183.2:5060;branch=z9hG4bK661C27
From: "Leonardo Added" <sip:3711@10.115.183.2>;tag=AD3FD90-2E0
To: <sip:2430600@10.115.182.4>
Date: Mon, 11 Jan 2016 23:26:01 GMT
Call-ID: 82D9FFE1-B7F111E5-8281940F-54A652D1@10.115.183.2
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1452554766
Reason: Q.850;cause=38
Content-Length: 0

INVITE RECEIVED FROM CUCM

*Jan 11 23:26:01.487: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:92430600@10.6.7.2:5060 SIP/2.0
Via: SIP/2.0/TCP 10.5.196.6:5060;branch=z9hG4bK113d50d65246721
From: "Leonardo Added" <sip:3711@10.5.196.6>;tag=81706880~80f843ae-9636-4957-95d8-0e16860abb0e-24211283
To: <sip:92430600@10.6.7.2>
Date: Mon, 11 Jan 2016 23:25:06 GMT
Call-ID: 8af73a80-694139d2-d6a80f-6c4050a@10.5.196.6
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 2331458176-0000065536-0000466759-0113509642
Session-Expires:  1800
P-Asserted-Identity: "Leonardo Added" <sip:3711@10.5.196.6>
Remote-Party-ID: "Leonardo Added" <sip:3711@10.5.196.6>;party=calling;screen=yes;privacy=off
Contact: <sip:3711@10.5.196.6:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 233

Can you please change this to UDP on CUCM through sip security profile  and make another test call.

Br,

Nadeem Ahmed

[Leonardo Santana]

Hello Nadeem

I change the SIP Profile to UDP and now after 20 seconds the call is dropped.

Follow attached the debugs.

Regards

Leonardo Santana

[Nadeem Ahmed]

aaah!! its seems like its moving somewhere. Can you please collect below the debugs and let me see if there is any session expiration issue or something else.

debug voice ccapi inout

debug ip tcp transcation

debug ccsip message

Br,

Nadeem Ahmed

[Leonardo Santana]

Hello Nadeem,

Follow attached.

Regards

Leonardo Santana

[Dennis Mink]

not necessarily an answer, but 

that debug shows 7 or 8 SIP OK's going from your gateway to cucm, none of which get responded by an ACK coming from CUCM.   the after these 7 OK's the GW sends a BYE to your service provider. 

[Leonardo Santana]

What could be the reason to CUCM no responding this requisition?

Regards

[Vivek Batra]

Hi Leo,

Seems some issue with the SIP bind interface. Please check the following;

INVITE sip:92430600@10.6.7.2:5060 SIP/2.0

Above is the INVITE received from CUCM. This means 10.6.7.2 is configured as CUBE address on SIP Trunk page in CUCM.

o=CiscoSystemsSIP-GW-UserAgent 3127 5758 IN IP4 10.115.183.2

Above is the 200 OK sent by CUBE to CUCM. Please note that it is sourced from Telco facing interface viz 10.115.183.2. CUCM won't be having route this IP and hence not able to send ACK. CUBE keep re transmitting 200 OK and then drop the call.

Please associate the correct SIP bind interface on inbound dial-peer and share the results.

- Vivek

[Leonardo Santana]

Yes,

How can i send all the outbound calls with the ip address off 10.115.183.2?

Because if change this will not work because the ITSP only accept calls from 10.115.183.2.

Regards

Leonardo Santana

[Vivek Batra]

Hi Leo,

I am not proposing to change anything on ITSP facing interface. This seems fine because all packets sent to ITSP are sourced via 10.115.183.2.

Issue is with between CUCM and CUBE. From the available logs, it seems you would like to use 10.6.7.2 interface between CUCM and CUBE. If that is the case, bind the incoming dial-peer (when making call from CUCM to PSTN) with 10.6.7.2 interface.

Also share the full config including all interfaces, dial-peers and translations. I would be better suggest on making desired changes.

- Vivek

[Leonardo Santana]

Hello Vivek,

Follow my config

interface GigabitEthernet0/0.230
description description ----------"[VOICE]"
encapsulation dot1Q 230
ip address 10.6.7.2 255.255.255.128

interface GigabitEthernet0/1
description -------------"[SIP TRUNK]"
ip address 10.115.183.2 255.255.255.252
duplex auto
speed auto

voice service voip
ip address trusted list
ipv4 10.6.7.0 255.255.255.128
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server

dial-peer voice 9001 voip
description -------------[dialplan-to-CCM-01]
preference 1
destination-pattern 86..$
session protocol sipv2
session target ipv4:10.5.196.6
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.230
voice-class sip bind media source-interface GigabitEthernet0/0.230
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 9002 voip
description -------------[dialplan-to-CCM-02]
preference 2
destination-pattern 86..$
session protocol sipv2
session target ipv4:10.5.196.7
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.230
voice-class sip bind media source-interface GigabitEthernet0/0.230
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 1 voip
description -------------[IN DDRs CUCM]
translation-profile incoming InboundPSTN
session protocol sipv2
session target ipv4:10.115.182.4
incoming called-number .T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 2 voip
description -------------[LOCAL-CEL]
translation-profile outgoing PSTNAccess
destination-pattern 9[2-8]......
session protocol sipv2
session target ipv4:10.115.182.4
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad
!
dial-peer voice 3 voip
description -------------[EMERGENCE]
translation-profile outgoing PSTNAccess
destination-pattern 9911
session protocol sipv2
session target ipv4:10.115.182.4
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad
!
dial-peer voice 4 voip
description -------------[DDI]
translation-profile outgoing PSTNAccess
destination-pattern 900T
session protocol sipv2
session target ipv4:10.115.182.4
session transport udp
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad

Thanks

[Vivek Batra · Accepted solution]

In the previous configuration, dial-peer 1 was bound with gig 0/1 interface but seems now you have changed it to gig 0/0.230 which seems fine for outgoing calls. But it will create issue with incoming calls from ITSP, as you have only one dial peer with incoming called-number command.

You need to have two separate inbound dial peers to handle incoming call legs from CUCM and ITSP.

For inbound dial peer from CUCM to CUBE, bind with gig 0/0.230. For inbound dial peer from ITSP to CUBE, bind with gig 0/1.

- Vivek

[Leonardo Santana]

Now its working is configured this way.

The previous post was wrong now its correct

dial-peer voice 9001 voip
description -------------[dialplan-to-CCM-01]
preference 1
destination-pattern 86..$
session protocol sipv2
session target ipv4:10.5.196.6
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.230
voice-class sip bind media source-interface GigabitEthernet0/0.230
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 9002 voip
description -------------[dialplan-to-CCM-02]
preference 2
destination-pattern 86..$
session protocol sipv2
session target ipv4:10.5.196.7
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.230
voice-class sip bind media source-interface GigabitEthernet0/0.230
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 1 voip
description -------------[IN DDRs CUCM]
translation-profile incoming InboundPSTN
session protocol sipv2
session target ipv4:10.115.182.4
incoming called-number .T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0.230
voice-class sip bind media source-interface GigabitEthernet0/0.230
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
!
dial-peer voice 2 voip
description -------------[LOCAL-CEL]
translation-profile outgoing PSTNAccess
destination-pattern 9[2-8]......
session protocol sipv2
session target ipv4:10.115.182.4
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad
!
dial-peer voice 3 voip
description -------------[EMERGENCE]
translation-profile outgoing PSTNAccess
destination-pattern 9911
session protocol sipv2
session target ipv4:10.115.182.4
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad
!
dial-peer voice 4 voip
description -------------[DDI]
translation-profile outgoing PSTNAccess
destination-pattern 900T
session protocol sipv2
session target ipv4:10.115.182.4
session transport udp
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate disable
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
icpif 50
no vad

But if tried to put on hold, transfer or make a conf the call is dropped

Follow the debug

[Nadeem Ahmed]

Leonardo,

Can you just test one more thing meanwhile i am deep diving with logs. enable the PRACK on CUBE and CUCM

http://www.cisco.com/c/en/us/support/docs/voice/session-initiation-protocol-sip/116086-configure-cube-cucm-sip-00.html

Global level:

voice service voip
  sip 
   rel1xx require 100rel

Dial-peer level:

dial-peer voice 1000 voip
  voice-class sip rel1xx require 100rel

[Leonardo Santana]

Hello i enabled at CUCM and at the Voice gateway.

I made a test and a receive this error:

SIP/2.0 500 Internal Server Error

Reason: Q.850;cause=127

Debus attached

Regards