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New Member

Outside call from CUCM not going

Hi,

I am using Cisco CUCM 7.1.5 which is connected to Cisco Voice gateways 2911 via SIP trunk with by default settings. Everything is working fine.

Now due to some reason, i have add one more third part SIP/GSM gateway on my LAN. So i have configured below dial-peer in order to route some desired voice traffic from that SIP/GSM gateway.

dial-peer voice 275 voip

description *** Router to 2n ***

destination-pattern 900T

session protocol sipv2

session target ipv4:X.X.X.X

voice-class codec 1

dtmf-relay sip-kpml sip-notify

no vad

Now IP Phones registered with CUCM when dial outside using the above mentioned dial-peer, call connects (as it shows on CDR logs on SIP/GSM gateway) but problem is that both part cant listen to each other.

What could be the possible reason? thanks

Message was edited by: Omer Tasaddaq

21 REPLIES
New Member

Re: Outside call from CUCM not going

Anyone here to give clue....


Sent from Cisco Technical Support iPad App

Re: Outside call from CUCM not going

Does your inbound call works from that new gsm/sip gateway ?

Sent from Cisco Technical Support iPhone App

New Member

Re: Outside call from CUCM not going

Let me tell you...phones registered in cme (2911) Cisco gateway dial outside and it does work perfect. I didn't check inside call.

Sent from Cisco Technical Support iPad App

Outside call from CUCM not going

Hi,

It seems routing issue between the IP address.

did you check the binding interface for media and control?

can you collect debug ccsip mess for the faulty call and provide the running config of SIP GW.

btw is it CME or CUCM7.1? could you please provide the complete call flow clearly?

//Suresh Please rate all the useful posts.
New Member

Re: Outside call from CUCM not going

This Cisco router is working as a cme-srst router.

Sent from Cisco Technical Support iPhone App

Outside call from CUCM not going

Hi.

Please attach the output of a debug ccsip messages during a non.working call.

Thanks

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"
Cisco Employee

Outside call from CUCM not going

Hi Omerpal,

Since you are seeing issues after the call connect, this means that

1. The communication IP address on the 200 OK and ACK messages is not right.

2. The communication IP address on the above 2 messages is not reachable. Packets are not being routed correctly.

Ideally, the RTP should be flowing in the following manner unless there is an MTP invoked for this call.

1. Between this router and IP Phone.

2. Between this router and far end.

The best way to troubleshoot this would be to take debug ccsip message on the router, as well as packet captures on the LAN facing interface, that would show you whether packets are leaving this router toward the phone/MTP. Also, you need to take packet captures on the IP Phone to see if the packets from the router are making it here, and packets from the phone are leaving it and reaching the router. If either of these packet captures indicates packet drops or missing streams, we know there is a network issue.

Thanks

Sreekanth

New Member

Re: Outside call from CUCM not going

IP Address of CUCM=10.50.2.3 & 10.50.2.4

IP Address of CME=10.1.184.65 (Connected with CUCM Via SIP Trunk)

IP Adress of SIP/GSM Gateway=10.1.163.130

To: <900923006064393>;tag=5DE115C0-1311

Date: Thu, 26 Dec 2013 13:33:16 GMT

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 103 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=0,OS=0,PR=984,OR=19680,PL=0,JI=0,LA=0,DU=20

Content-Length: 0

Dec 26 13:33:17.003: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SUBSCRIBE sip:3916@10.50.2.4:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.184.65:5060;branch=z9hG4bK107B181F

From: <900923336064393>;tag=5DE115C0-1311

To: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e77d

-38640527

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 102 SUBSCRIBE

Max-Forwards: 70

Date: Thu, 26 Dec 2013 13:33:17 GMT

User-Agent: Cisco-SIPGateway/IOS-12.x

Event: kpml

Expires: 0

Contact: <10.1.184.65:5060>

Content-Length: 0

Dec 26 13:33:17.003: //5905/040BAB038759/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:900923336064393@10.1.163.130:5060;transport=TCP SIP/2.0

Via: SIP/2.0/TCP 10.1.184.65:5060;branch=z9hG4bK107A161C

From: "Ammar Javaid" <3916>;tag=5DE0F848-369

To: <900923336064393>;tag=7c1eb3fe1214-128-1a102f20

Date: Thu, 26 Dec 2013 13:32:35 GMT

Call-ID: 40CE353-6D6911E3-875F90DE-E66116E9@10.1.184.65

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1388064797

CSeq: 102 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=984,OS=19680,PR=0,OR=0,PL=0,JI=0,LA=0,DU=20

Content-Length: 0

Dec 26 13:33:17.491: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:900923336064393@10.1.184.65:5060 SIP/2.0

Date: Thu, 26 Dec 2013 13:38:04 GMT

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

P-Asserted-Identity: "Ammar Javaid" <3916>

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <900923336064393>;tag=5DE115C0-1311

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720942969b75b

CSeq: 103 BYE

Max-Forwards: 70

Dec 26 13:33:17.491: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720942969b75b

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

To: <900923336064393>;tag=5DE115C0-1311

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 103 BYE

Content-Length: 0

Dec 26 13:33:17.503: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SUBSCRIBE sip:3916@10.50.2.4:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.184.65:5060;branch=z9hG4bK107B181F

From: <900923336064393>;tag=5DE115C0-1311

To: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e77d

-38640527

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 102 SUBSCRIBE

Max-Forwards: 70

Date: Thu, 26 Dec 2013 13:33:17 GMT

User-Agent: Cisco-SIPGateway/IOS-12.x

Event: kpml

Expires: 0

Contact: <10.1.184.65:5060>

Content-Length: 0

Dec 26 13:33:17.687: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

NOTIFY sip:10.1.184.65:5060 SIP/2.0

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

P-Asserted-Identity: "Ammar Javaid" <3916>

Event: kpml

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <900923336064393>;tag=5DE115C0-1311

Contact: <3916>

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Subscription-State: terminated

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720957b6a3021

CSeq: 104 NOTIFY

Max-Forwards: 70

Dec 26 13:33:17.691: //0/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: <900923336064393>;tag=5DE115C0-1311

Content-Length: 0

To: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e77d

-38640527

Contact: <3916>

Expires: 0

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Via: SIP/2.0/UDP 10.1.184.65:5060;branch=z9hG4bK107B181F

CSeq: 102 SUBSCRIBE

Dec 26 13:33:17.691: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

NOTIFY sip:10.1.184.65:5060 SIP/2.0

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

P-Asserted-Identity: "Ammar Javaid" <3916>

Event: kpml

Content-Length: 348

User-Agent: Cisco-CUCM7.1

To: <900923336064393>;tag=5DE115C0-1311

Contact: <3916>

Content-Type: application/kpml-response+xml

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Subscription-State: terminated;reason=timeout

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK72096181a6806

CSeq: 105 NOTIFY

Max-Forwards: 70

http://ww

w.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpm

l-response kpml-response.xsd" code="487" digits="" forced_flush="false" suppress

ed="false" tag="dtmf" text="Subscription Expired" version="1.0"/>

Dec 26 13:33:17.691: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720957b6a3021

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

To: <900923336064393>;tag=5DE115C0-1311

Date: Thu, 26 Dec 2013 13:33:17 GMT

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 104 NOTIFY

Content-Length: 0

Dec 26 13:33:17.691: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 489 Bad Event - 'Malformed/Unsupported Event'

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK72096181a6806

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

To: <900923336064393>;tag=5DE115C0-1311

Date: Thu, 26 Dec 2013 13:33:17 GMT

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 105 NOTIFY

Allow-Events: kpml, telephone-event

Content-Length: 0

Dec 26 13:33:18.179: //0/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: <900923336064393>;tag=5DE115C0-1311

Content-Length: 0

To: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e77d

-38640527

Contact: <3916>

Expires: 0

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Via: SIP/2.0/UDP 10.1.184.65:5060;branch=z9hG4bK107B181F

CSeq: 102 SUBSCRIBE

Dec 26 13:33:18.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

NOTIFY sip:10.1.184.65:5060 SIP/2.0

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

P-Asserted-Identity: "Ammar Javaid" <3916>

Event: kpml

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: <900923336064393>;tag=5DE115C0-1311

Contact: <3916>

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Subscription-State: terminated

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720957b6a3021

CSeq: 104 NOTIFY

Max-Forwards: 70

Dec 26 13:33:18.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

NOTIFY sip:10.1.184.65:5060 SIP/2.0

Date: Thu, 26 Dec 2013 13:38:45 GMT

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

P-Asserted-Identity: "Ammar Javaid" <3916>

Event: kpml

Content-Length: 348

User-Agent: Cisco-CUCM7.1

To: <900923336064393>;tag=5DE115C0-1311

Contact: <3916>

Content-Type: application/kpml-response+xml

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

Subscription-State: terminated;reason=timeout

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK72096181a6806

CSeq: 105 NOTIFY

Max-Forwards: 70

http://ww

w.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpm

l-response kpml-response.xsd" code="487" digits="" forced_flush="false" suppress

ed="false" tag="dtmf" text="Subscription Expired" version="1.0"/>

Dec 26 13:33:18.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK720957b6a3021

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

To: <900923336064393>;tag=5DE115C0-1311

Date: Thu, 26 Dec 2013 13:33:17 GMT

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 104 NOTIFY

Content-Length: 0

Dec 26 13:33:18.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 489 Bad Event - 'Malformed/Unsupported Event'

Via: SIP/2.0/UDP 10.50.2.4:5060;branch=z9hG4bK72096181a6806

From: "Ammar Javaid" <3916>;tag=950e340e-4e22-4901-ae4b-70c24364e7

7d-38640527

To: <900923336064393>;tag=5DE115C0-1311

Date: Thu, 26 Dec 2013 13:33:17 GMT

Call-ID: f0427500-2bc1313b-330ee-402320a@10.50.2.4

CSeq: 105 NOTIFY

Allow-Events: kpml, telephone-event

Content-Length: 0

Dec 26 13:33:23.939: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:3916@10.1.184.65:5060 SIP/2.0

Max-Forwards: 70

Via: SIP/2.0/TCP 10.1.163.130:5060;branch=z9hG4bK-75437268499

From: <900923336064393>;tag=7c1eb3fe1214-128-1a102f20

To: "Ammar Javaid" <3916>;tag=5DE0F848-369

Call-ID: 40CE353-6D6911E3-875F90DE-E66116E9@10.1.184.65

CSeq: 69 BYE

Contact: <900923336064393>

User-Agent: 2N VoiceBlueNext 1.5.1.7.2

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY

Content-Length: 0

Dec 26 13:33:23.943: //5905/040BAB038759/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.1.163.130:5060;branch=z9hG4bK-75437268499

From: <900923336064393>;tag=7c1eb3fe1214-128-1a102f20

To: "Ammar Javaid" <3916>;tag=5DE0F848-369

Date: Thu, 26 Dec 2013 13:33:23 GMT

Call-ID: 40CE353-6D6911E3-875F90DE-E66116E9@10.1.184.65

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 69 BYE

P-RTP-Stat: PS=984,OS=19680,PR=0,OR=0,PL=0,JI=0,LA=0,DU=27

Content-Length: 0

Cisco Employee

Re: Outside call from CUCM not going

Hi Omerpal,

These logs don't seem complete. They do not show the entire call. Please take the debug ccsip message again and record it to the router buffer first. You can then use the show logging command to get the output.

Thanks

Sree

New Member

Re: Outside call from CUCM not going

debug ccsip error:

001264: Dec 28 07:55:17.213: //5992/146BCA31885F/SIP/Error/ccsip_ipipms_peer_chnl_ind_hdlr: Peer caps is NULL

001265: Dec 28 07:55:20.361: //5992/146BCA31885F/SIP/Error/ccsip_alert_parent_or_child: Unable to add unsupported

                                      hdrs to container

SIP: Attribute mid, level 1 instance 1 not found.

SIP: Attribute ptime, level 1 instance 1 not found.

SIP: Attribute ptime, level 1 instance 1 not found.

001266: Dec 28 07:55:32.141: //-1/xxxxxxxxxxxx/SIP/Error/sipSPICheckAndClearSrcSRTPSdp: CCB SDP source pointer NULL

SIP: (5991) Group (a= group line) attribute, level 65535 instance 1 not found.

SIP: (5991) Group (a= group line) attribute, level 65535 instance 1 not found.

001267: Dec 28 07:55:32.657: //5992/146BCA31885F/SIP/Error/sact_recd_success_new_message_response: Unexpected response

SIP: Attribute mid, level 1 instance 1 not found.

001268: Dec 28 07:55:33.049: //5991/146BCA31885F/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS params for midcall INVITE

001269: Dec 28 07:55:33.049: //5992/146BCA31885F/SIP/Error/sipSPIHandleSDPOwnerVersionIDChange: Pointers are NULL..

debug ccsip calls:

001271: Dec 28 07:58:21.153: //5993/797989178866/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31FB9B40

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : NO

Calling Number           : 2233

Called Number            : 900923008458979

Source IP Address (Sig  ): 10.1.184.65

Destn SIP Req Addr:Port  : 10.50.2.4:5060

Destn SIP Resp Addr:Port : 10.50.2.4:5060

Destination Name         : 10.50.2.4

001272: Dec 28 07:58:21.153: //5993/797989178866/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.184.65

Source IP Port    (Media): 29502

Destn  IP Address (Media): 10.1.18.48

Destn  IP Port    (Media): 21754

Orig Destn IP Address:Port (Media): [ - ]:0

001273: Dec 28 07:58:21.153: //5993/797989178866/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31FB9B40

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : NO

Calling Number           : 2233

Called Number            : 900923008458979

Source IP Address (Sig  ): 10.1.184.65

Destn SIP Req Addr:Port  : 10.50.2.4:5060

Destn SIP Resp Addr:Port : 10.50.2.4:5060

Destination Name         : 10.50.2.4

001274: Dec 28 07:58:21.153: //5993/797989178866/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.184.65

Source IP Port    (Media): 29502

Destn  IP Address (Media): 10.1.18.48

Destn  IP Port    (Media): 21754

Orig Destn IP Address:Port (Media): [ - ]:0

001275: Dec 28 07:58:21.157: //5994/797989178866/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31F9D990

State of The Call        : STATE_ACTIVE

TCP Sockets Used         : YES

Calling Number           : 2233

Called Number            : 900923008458979

Source IP Address (Sig  ): 10.1.184.65

Destn SIP Req Addr:Port  : 10.1.163.130:5060

Destn SIP Resp Addr:Port : 10.1.163.130:5060

Destination Name         : 10.1.163.130

001276: Dec 28 07:58:21.157: //5994/797989178866/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.184.65

Source IP Port    (Media): 18256

Destn  IP Address (Media): 10.1.163.130

Destn  IP Port    (Media): 8266

Orig Destn IP Address:Port (Media): [ - ]:0

001277: Dec 28 07:58:40.565: //5994/797989178866/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31F9D990

State of The Call        : STATE_DEAD

TCP Sockets Used         : YES

Calling Number           : 2233

Called Number            : 900923008458979

Source IP Address (Sig  ): 10.1.184.65

Destn SIP Req Addr:Port  : 10.1.163.130:5060

Destn SIP Resp Addr:Port : 10.1.163.130:4159

Destination Name         : 10.1.163.130

001278: Dec 28 07:58:40.565: //5994/797989178866/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.184.65

Source IP Port    (Media): 18256

Destn  IP Address (Media): 10.1.163.130

Destn  IP Port    (Media): 8266

Orig Destn IP Address:Port (Media): [ - ]:0

001279: Dec 28 07:58:40.565: //5994/797989178866/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 200

001280: Dec 28 07:58:41.241: //5993/797989178866/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31FB9B40

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 2233

Called Number            : 900923008458979

Source IP Address (Sig  ): 10.1.184.65

Destn SIP Req Addr:Port  : 10.50.2.4:5060

Destn SIP Resp Addr:Port : 10.50.2.4:5060

Destination Name         : 10.50.2.4

001281: Dec 28 07:58:41.241: //5993/797989178866/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.184.65

Source IP Port    (Media): 29502

Destn  IP Address (Media): 10.1.18.48

Destn  IP Port    (Media): 21754

Orig Destn IP Address:Port (Media): [ - ]:0

001282: Dec 28 07:58:41.241: //5993/797989178866/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 200

Message was edited by: Omer Tasaddaq

New Member

Re: Outside call from CUCM not going

Anyone here to give clue....

Outside call from CUCM not going

Hi Omar,

From the SIP messages exchanged between CUCM, CME & GSM GW, we see the below.

Between CUCM & CME

CUCM media: c=IN IP4 10.1.18.48 codec: g729 (think this this cisco phone ip addr)

CME media: c=IN IP4 10.1.184.65 codec: g729

between CME & GSM

GSM Media: c=IN IP4 10.1.163.130 codec: g729

CME Media: c=IN IP4 10.1.184.65  codec: g729

and if I understand correctly, the call flow is:

non working call flow: IP phone----CUCM----SIP----CME---GSM-GW----PSTN

working flow : IP phone---CME---GSM-GW----PSTN

in that case, we need to check how the CME is integrated with CUCM, pls check binding interface in cme for media and signaling

could you please provide the CME Running Configuration?

//Suresh Please rate all the useful posts.
New Member

Re: Outside call from CUCM not going

Thanks for your reply.

Call flow is....

IP Phone---CUCM---SIP Trunk---CME---SIP Trunk---2n (SIP/GSM Gateway)---GSM---PSTSN

where 2n is 3rd part SIP & GSM gateway. Its working with H.323 trunk (between CUCM & CME). This time we have 2 SIP Trunks in call flow.

Please find below CME config.

voice-card 0

dspfarm

dsp services dspfarm

!

!

!

voice service voip

media flow-around

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

sip

  bind control source-interface Loopback105

  bind media source-interface Loopback105

  registrar server

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class custom-cptone 2n-gsm

dualtone disconnect

  frequency 425

  cadence 330 330

!

!

voice iec syslog

!

!

interface Loopback105

ip address 10.1.184.65 255.255.255.192

!

sccp local Loopback105

sccp ccm 10.50.2.3 identifier 1 version 7.0

sccp ccm 10.50.2.4 identifier 2 version 7.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 3 register Al_MTP

associate profile 2 register Al_XCODE

associate profile 1 register Al_CONF

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 12

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 6

associate application SCCP

!

dspfarm profile 3 mtp

codec g711ulaw

maximum sessions software 10

associate application SCCP

!

dial-peer voice 275 voip

description *** test ***

destination-pattern 900T

session protocol sipv2

session target ipv4:10.1.163.130

dtmf-relay sip-kpml sip-notify

codec g711alaw

no vad

!

Re: Outside call from CUCM not going

could you please include the dial-peer configuration to/from CUCM? it would be great if you could post the complete running config.

also, are you able to ping CUCM or Cisco Phone's IP address from GSM/SIP GW?

Message was edited by: Suresh Subramanian

//Suresh Please rate all the useful posts.
New Member

Re: Outside call from CUCM not going

Actually there is a very list of username i have added in cme, then ephone and ephone-dn.

thats why i have sent to you the relative info.

dial-peer voice 100 voip

destination-pattern [1-8]...$

session protocol sipv2

session target ipv4:10.50.2.3

incoming called-number .

dtmf-relay sip-kpml sip-notify

no vad

!

dial-peer voice 101 voip

destination-pattern [1-8]...$

session protocol sipv2

session target ipv4:10.50.2.4

incoming called-number .

dtmf-relay sip-kpml sip-notify

no vad

Outside call from CUCM not going

if you call the IP phone registered in CME from the IP phone registered in CUCM, do you get two way audio?

if you call the GSM number from CME IP phone thru GSM SIP GW, do you get the two way audio?

for testing purpose, could you please try enabling  MTP check box in SIP Trunk in CUCM and test the GSM calls?

what is this IP address: 10.1.18.48 ?

from GSM GW, are you able to ping the IP  10.1.18.48 and CUCM IP addresses?

//Suresh Please rate all the useful posts.
New Member

Re: Outside call from CUCM not going

if you call the IP phone registered in CME from the IP phone registered in CUCM, do you get two way audio? YES

if you call the GSM number from CME IP phone thru GSM SIP GW, do you get the two way audio? YES

for testing purpose, could you please try enabling  MTP check box in SIP Trunk in CUCM and test the GSM calls? I'll do now

what is this IP address: 10.1.18.48 ? Its my CIscp IP Phone.

IP Address of CUCM=10.50.2.3 & 10.50.2.4

IP Address of CME=10.1.184.65 (Connected with CUCM Via SIP Trunk)

IP Adress of SIP/GSM Gateway=10.1.163.130

from GSM GW, are you able to ping the IP  10.1.18.48 and CUCM IP addresses? YES

New Member

Re: Outside call from CUCM not going

if i enable MTP checkbox in CUCM trunk page....after that when i make call it gives me immediately fast busy tone.

New Member

Re: Outside call from CUCM not going

Anyone.....

Outside call from CUCM not going

Hi Omer, is there any specific reason behind to have the CME in between CUCM & GSM GW?

why don't you integrate the GW with CUCM directly using SIP Trunk?

BYE message sent from CME to CUCM with packet statistics

------------------------------------------------------------------------------------------

002737: Dec 28 08:35:54.489: //6005/AA2FA7EB888B/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:2233@10.50.2.4:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.184.65:5060;branch=z9hG4bK109214A0

From: <900924237802387>;tag=6678BA18-1E0A

To: "Omer Tasaddaq" <2233>;tag=950e340e-4e22-4901-ae4b-70c24364e77d-38662205

Date: Sat, 28 Dec 2013 05:35:46 GMT

Call-ID: 9781f200-2be1645d-3329c-402320a@10.50.2.4

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1388208954

CSeq: 101 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=0,OS=0,PR=412,OR=8240,PL=0,JI=0,LA=0,DU=9

>> PR=412; 412 packets received on this call leg but no packet sent (PS=0)

BYE message received from GSM GW to CME

---------------------------------------------------------------------

002634: Dec 28 08:35:53.985: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:2233@10.1.184.65:5060 SIP/2.0

Max-Forwards: 70

Via: SIP/2.0/TCP 10.1.163.130:5060;branch=z9hG4bK-82470299394

From: <900924237802387>;tag=7c1eb3fe1214-135-1c083219

To: "Omer Tasaddaq" <2233>;tag=6678ADE8-85B

Call-ID: AA321924-6EB811E3-889190DE-E66116E9@10.1.184.65

CSeq: 76 BYE

Contact: <900924237802387>

User-Agent: 2N VoiceBlueNext 1.5.1.7.2

Allow: INVITE,BYE,ACK,CANCEL,OPTIONS,REFER,NOTIFY

Content-Length: 0

200OK sent back to GSM GW with packet statistics

------------------------------------------------------------------------------

002726: Dec 28 08:35:53.993: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 10.1.163.130:5060;branch=z9hG4bK-82470299394

From: <900924237802387>;tag=7c1eb3fe1214-135-1c083219

To: "Omer Tasaddaq" <2233>;tag=6678ADE8-85B

Date: Sat, 28 Dec 2013 05:35:53 GMT

Call-ID: AA321924-6EB811E3-889190DE-E66116E9@10.1.184.65

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 76 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=412,OS=8240,PR=0,OR=0,PL=0,JI=0,LA=0,DU=9

>> sent 412 packets and no packets received

>> with this statistics, it seems oneway audio issue rather than no way audio.

>> perhaps we need to do the packet capturing by spanning the switch ports where the IP Phone & GSM GW are connected to

>> in the IP phone, if you press '?' button twice, you will also get the RTP statistics. could you please try that and check whether or not the IP phone sending/receiving the packets

//Suresh Please rate all the useful posts.
New Member

Outside call from CUCM not going

Hi omerpal1190,

Your problem was solved?

In my Lab I could make it work!

Regards,

Bruno

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