CUCM Lab - VGW - SIP Trunk Service Provider Problem

[born.jason]

Hi,

setting up a lab with cucm and a h323 gateway. Diealpeers fine. Internal Calls works well. But i have problems with outbound and inbound calls. I can dial in or make a call out but if the other side respective the internal side take the call i got after a few seconds a busy tone.

Any suggestions?

Here is a debug ccsip messages after the dial tone appears:

BYE sip:23004976212345678@217.10.77.44:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.99:5060;branch=z9hG4bK8F1BF1

From: <sip:13XXXXX@sipgate.de>;tag=4A4298-176E

To: <sip:004976212345678@sipgate.de>;tag=as406f85b3

Date: Fri, 02 Mar 2012 21:23:17 GMT

Call-ID: C343C0D3-63E411E1-8047D687-2AF204BE@sipgate.de

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

Max-Forwards: 70

Route: <sip:217.10.79.9;lr;ftag=4A4298-176E>,<sip:172.20.40.4;lr=on>

Timestamp: 1330723413

CSeq: 103 BYE

Reason: Q.850;cause=47

Proxy-Authorization: Digest username="13XXXXX",realm="sipgate.de",uri="sip:23004976212345678@217.10.77.44:5060",response="a93da886a1a282937b813548c6edc02d",nonce="4f513bda3fe03ecd12fb35f2b3d1e83084b7b1dd",algorithm=md5

P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=12

Content-Length: 0

Regards

Jason

[Vipul Jindal]

cause code

Reason: Q.850;cause=47

is for media resrouce issue.

you may need to check the region and location config. if that is ok we need to take a look at traces to check what going on.

thanks

Vipul Jindal

[born.jason]

Hi Vipul,

Regions and location looks fine i think. I have tried the default and customized. What do you think should we do next to troubleshoot?

Regards

Jason

[born.jason]

Ok, it goes ahead.

1st problem:

I deleted the voice class codec andcreated it new and for that reason the voice-class codec 1 command went away from the dial peers, i now re added it. Now i`m able to perform an outgoing call from the phone BUT WITHOUT dial tone. If the call is established i can hear the called number and he me.

Any suggestions?

UPDATE:

Hmm to some numbers it works with dial tone. I have make a debug ccsip calls two both working dial tone and not working dial tone and see the folowing:

NOT working DIAL TONE:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711ulaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 0 (tx), 0 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 99 (tx), 99 (rx)

Source IP Address (Media): 192.168.5.99

Source IP Port    (Media): 18682

Destn  IP Address (Media): 62.53.226.17

Destn  IP Port    (Media): 19578

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

WORKING DIAL TONE:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711ulaw

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 0 (tx), 0 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 192.168.5.99

Source IP Port    (Media): 17344

Destn  IP Address (Media): 62.52.147.156

Destn  IP Port    (Media): 22634

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

the destination ip is different and the negotiated code Bytes. Btw. on the working number i can also make Music on Hold. If i make this on the other not working number the call disconnects.

Suggestions for that problem?

2nd problem:

for incoming calls the phone rings but if i catch the call, the phone on the other side don`t noticed that and he hear a dial tone as nobody pick up the phone. So the call could not be established.

Any suggestions?

Hope somebody could help. If you need any debugs from  voice gw or cucm please let me know.

regards

Jason

[Vipul Jindal]

i guess i need to take a look at the call manager logs to check whats going on!1

thanks,

Vipul JIndal

[born.jason]

Which logs (from RTMT?) do you need exactly?

Thanks

Jason

[Vipul Jindal]

from RTMT get cisco call manager logs!!

thanks,

Vipul Jindal