Solved: Hi Aseem,

Sanjay Bishnoi · ‎09-29-2016

SIP trunk not getting register with telco and my scenario is CUCM>CUBE GATEWAY>SP GATEWAY>SBC server

Kindly suggest me how can i get register with service provider

Aseem Anand · ‎09-29-2016

Hi Sanjay,

Seems to be a TCP issue. Can you try changing the transport type to UDP on the trunk under security profile and on the incoming dial-peer and see if it helps?

Also make sure you have the correct address binded on gateway and correct destination address specified on the trunk.

Aseem

View solution in original post

Aseem Anand · ‎09-29-2016

Hi Sanjay,

I just checked your running configuration and found a sip-profile which is modifying the SIP URI. Not sure why you are using this but can you remove the SIP profile from dial-peer 3 and check if the call works?

voice class sip-profiles 2
 request INVITE peer-header sip TO copy "sip:(.*)@" u01 
 request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1" 
!

Also if you can help me understand the reason for this SIP profile. What are you actually trying to change in the SIP URI?

Aseem

(Please rate if it helps)

View solution in original post

Aseem Anand · ‎09-29-2016

Hi Sanjay,

Can you please share your sip-ua configuration? You will need a username password provided by the provider to register with the provider. Also, make sure you are able to resolve the hostname/FQDN of the provider if ip address is not used.

Please go through the link below to understand the integration:

http://www.cisco.com/c/en/us/td/docs/ios/voice/sip/configuration/guide/15_1/sip_15_1_book/sip_cg-basic_cfg.pdf

Aseem

(Please rate if useful)

Sanjay Bishnoi · ‎09-29-2016

Hello Aseem,

thanks for update ..

we are using SIP server IP address and find below SIP UA

sip-ua
credentials username 6253600 password 7 1234 realm 10.50.128.2
authentication username 6253600 password 7 1234
no remote-party-id
retry invite 5
retry register 5
retry options 10
timers connect 100
registrar 1 ipv4:10.50.128.2:5060 expires 3600
registration spike 50
host-registrar

Aseem Anand · ‎09-29-2016

Hi,

Can you please collect the debugs below:

debug ccsip events
debug ccsip messages

Aseem

Sanjay Bishnoi · ‎09-29-2016

hello Assem,

now good news that cucm get registered with ISP but still unable to make any outgoing and incoming calls. logs are below mentioned

NAINGRVG02#
NAINGRVG02#debug ccsip event
SIP Call events tracing is enabled
NAINGRVG02#
NAINGRVG02#
NAINGRVG02#debug ccsip e
000525: *Sep 29 12:18:17.458: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_OPTIONS_RESPssage
^
% Invalid input detected at '^' marker.

NAINGRVG02#
NAINGRVG02#debug cc
000526: *Sep 29 12:18:23.368: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_OPTIONS_RESPsip message
SIP Call messages tracing is enabled
NAINGRVG02#
NAINGRVG02#
NAINGRVG02#
NAINGRVG02#
000527: *Sep 29 12:18:48.683: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:909873305718@10.13.84.9:5060 SIP/2.0
Via: SIP/2.0/TCP 10.13.84.2:5060;branch=z9hG4bK25303974ca98
From: "Reception X 3600" <sip:87003600@10.13.84.2>;tag=15938~45279cfa-7302-8ca9-fc3c-b5dba437e2a2-27805257
To: <sip:909873305718@10.13.84.9>
Date: Thu, 29 Sep 2016 12:17:19 GMT
Call-ID: a966b700-7ed1064f-2419-2540d0a@10.13.84.2
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
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,X-cisco-original-called
Call-Info: <sip:10.13.84.2:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 2842081024-0000065536-0000000133-0039062794
Session-Expires: 1800
P-Asserted-Identity: "Reception X 3600" <sip:87003600@10.13.84.2>
Remote-Party-ID: "Reception X 3600" <sip:87003600@10.13.84.2>;party=calling;screen=yes;privacy=off
Contact: <sip:87003600@10.13.84.2:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="SEP00562B624077"
Max-Forwards: 69
Content-Length: 0

000528: *Sep 29 12:18:48.684: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
000529: *Sep 29 12:18:48.684: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:7F89EECC12F8
000530: *Sep 29 12:18:48.684: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:none, Min-SE Value:1800, flags:2001
000531: *Sep 29 12:18:48.686: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
000532: *Sep 29 12:18:48.686: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:7F89EECC13C8
000533: *Sep 29 12:18:48.687: //343/A966B7000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.13.84.2:5060;branch=z9hG4bK25303974ca98
From: "Reception X 3600" <sip:87003600@10.13.84.2>;tag=15938~45279cfa-7302-8ca9-fc3c-b5dba437e2a2-27805257
To: <sip:909873305718@10.13.84.9>
Date: Thu, 29 Sep 2016 12:18:48 GMT
Call-ID: a966b700-7ed1064f-2419-2540d0a@10.13.84.2
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S2
Content-Length: 0

000534: *Sep 29 12:18:48.688: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
000535: *Sep 29 12:18:48.689: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
000536: *Sep 29 12:18:48.690: //344/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
000537: *Sep 29 12:18:48.690: //344/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:7F89EECC1DF0
000538: *Sep 29 12:18:48.690: //344/A966B7000000/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:none, Min-SE Value:1800, flags:2001
000539: *Sep 29 12:18:53.691: %VOICE_IEC-3-GW: SIP: Internal Error (Socket error): IEC=1.1.186.7.7.4 on callID 344 GUID=A966B700000100000000008502540D0A
000540: *Sep 29 12:18:53.691: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
000541: *Sep 29 12:18:53.691: //344/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SPI_EVENT
000542: *Sep 29 12:18:53.692: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
000543: *Sep 29 12:18:53.693: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
000544: *Sep 29 12:18:53.693: //343/A966B7000000/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:503, container:7F89EECC08D0
000545: *Sep 29 12:18:53.693: //343/A966B7000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 10.13.84.2:5060;branch=z9hG4bK25303974ca98
From: "Reception X 3600" <sip:87003600@10.13.84.2>;tag=15938~45279cfa-7302-8ca9-fc3c-b5dba437e2a2-27805257
To: <sip:909873305718@10.13.84.9>;tag=6049EE-2651
Date: Thu, 29 Sep 2016 12:18:48 GMT
Call-ID: a966b700-7ed1064f-2419-2540d0a@10.13.84.2
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S2
Reason: Q.850;cause=38
Content-Length: 0

000546: *Sep 29 12:18:53.695: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:909873305718@10.13.84.9:5060 SIP/2.0
Via: SIP/2.0/TCP 10.13.84.2:5060;branch=z9hG4bK25303974ca98
From: "Reception X 3600" <sip:87003600@10.13.84.2>;tag=15938~45279cfa-7302-8ca9-fc3c-b5dba437e2a2-27805257
To: <sip:909873305718@10.13.84.9>;tag=6049EE-2651
Date: Thu, 29 Sep 2016 12:17:19 GMT
Call-ID: a966b700-7ed1064f-2419-2540d0a@10.13.84.2
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

Aseem Anand · ‎09-29-2016

Hi Sanjay,

Can you make sure you have the correct address bind for media & signalling on the gateway?

Make sure the destination you have defined in the trunk matches with the sip bind address on the CUBE?

Can you send me the running configuration from the CUBE along with the output of the debugs:

Debug voice ccapi inout

debug ccsip messages

Aseem

Sanjay Bishnoi · ‎09-29-2016

Hello Aseem,

Please find below attached run config and logs

calling number - 87003600 (DID - 124-6253600)

called number - 9873305718

Rajan · ‎09-29-2016

Hi Sanjay,

Once the gateway gets the Invite, it sends a trying and waits for 5 seconds before disconnecting the call with the below error due to TCP connection timeout:

000539: *Sep 29 12:18:53.691: %VOICE_IEC-3-GW: SIP: Internal Error (Socket error): IEC=1.1.186.7.7.4 on callID 344 GUID=A966B700000100000000008502540D0A

You could try increasing the timer using the below command as mentioned in this thread:

sip-ua

timers trying 100

http://www.gossamer-threads.com/lists/cisco/voip/184211

HTH

Rajan

Pls rate all useful posts

Aseem Anand · ‎09-29-2016

Hi Sanjay,

Seems to be a TCP issue. Can you try changing the transport type to UDP on the trunk under security profile and on the incoming dial-peer and see if it helps?

Also make sure you have the correct address binded on gateway and correct destination address specified on the trunk.

Aseem

Sanjay Bishnoi · ‎09-29-2016

Hello Aseem,

thanks for update and now my outgoing calls are working fine post change the session transport tcp to UDP.

but incoming calls are still not delivered..however TO HEADER to Request URI rule, it is copying u01 and not the full DID 1246253600

Instead of u01, router should populate 6253600 in the INVITE towards call manager

001009: *Sep 29 14:10:08.993: //693/436EAA0083EF/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:\u01@10.13.84.2:5060 SIP/2.0

Via: SIP/2.0/UDP 10.13.84.9:5060;branch=z9hG4bKDC2B8

From: "09873305718" <sip:9873305718@10.13.84.9>;tag=C62551-235D

To: <sip:87003600@10.13.84.2>

Date: Thu, 29 Sep 2016 14:10:08 GMT

Call-ID: 436F94D4-858511E6-83F5B3CD-D51B2B3A@10.13.84.9

Aseem Anand · ‎09-29-2016

Hi Sanjay,

I just checked your running configuration and found a sip-profile which is modifying the SIP URI. Not sure why you are using this but can you remove the SIP profile from dial-peer 3 and check if the call works?

voice class sip-profiles 2
 request INVITE peer-header sip TO copy "sip:(.*)@" u01 
 request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1" 
!

Also if you can help me understand the reason for this SIP profile. What are you actually trying to change in the SIP URI?

Aseem

(Please rate if it helps)

Sanjay Bishnoi · ‎09-29-2016

Hi Aseem,

thanks for your support and now my incoming and outgoing working fine.

but when i dial first time any number i get delay to deliver calls but when when i dial second time same number its delivered perfectly.

and DID is also not working as when dialed any DID number its delivered on pilot number

Unable to get register SIP trunk with service provider