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schoerlin
Level 1
Level 1

Hi all,

I did do a QSIG connection from a Cisco 2921 and a Siemens Hipath. All calls from the Siemens were transmitted over this QSIG . This QSIG connection was working without any issues. Without any configuration change, the Siemens devices behind the QSIG were not able any more, to do external calls. But they were able to do calls internally to the Cisco over the QSIG, and they were still reachable from externally.

If I try to do an external call on the Siemens, I get the following error message on the QSIG:

Aug 9 12:07:06.016 CET: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x806E
Aug 9 12:07:06.016 CET: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x806E
Cause i = 0x80AF - Resource unavailable, unspecified
Aug 9 12:07:06.076 CET: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x006E
Cause i = 0x8090 - Normal call clearing
Aug 9 12:07:06.164 CET: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x806E

I already exchanged the DSP`s on the QSIG, but unfortunately I got the same error.

Thanks in advance

Frank

1 Accepted Solution

Accepted Solutions

Can you disable "Early Offer support for voice and video calls" and check if that makes any difference.

Regards,

Mohammed Noor

View solution in original post

13 Replies 13

Mohammed Khan
Cisco Employee
Cisco Employee

Hi ,

can you  provide the following for a sample call

debug voip ccapi inout

debug isdn q931

debug dial-peer

show tech

show tech voice

Regards,

Mohammed Noor

Hi Mohammed,

many thanks for your quick reponse.

Attached the debugging of 3 calls, and the outputs of the show tech commands.

T&R

Frank

This is MGCP call flow. Need following logs from GW

debug mgcp events
debug mgcp packet
debug mgcp errors
debug vtsp events
debug vtsp session
debug voip ccapi inout
debug ccm-manager backhaul
debug isdn q931

Also with Detailed Cisco Callmanager Logs

Regards,

Mohammed Noor

Hi Mohammed,

here are the logs from the gateway.
Can you tell me which Call Manager Logs you exactly want? I think from the RTMT, but here you also have many options.

T&R

Frank

Follow below link to collect logs from RTMT

https://smbitsolutions.wordpress.com/2010/05/26/how-to-gather-logs-with-rtmt/

Regards,

Mohammed Noor

Hi Mohammed,

many thanks for the description. Attached the log file from the RTMT.

T&R

Frank

Logs don't cover failed call. You may have collect logs right after simulating a sample call.

Note:Collect logs from all server.

Regards,

Mohammed Noor

Hi Mohammed,

apologize for that.

Again all logs. I was able to find the call on the RTMT Log (13:37:55), therefore I hope you will see all necessary information.

Thanks in advance!

Frank

Frank

Call failing when it gets extended to SIP Trunk(DEAUUCG1). Can you send me the screenshot of SIP Trunk and SIP Profile configuration.

00311968.001 |13:38:15.572 |AppInfo  |SIPCdpc(219) - outCall_waitRSVPRes_PolicyAndCACRegisterRes: RSVP indicates the call to be cleared: RSVP_EO_RES_Fail_Call

Regards,

Mohammed Noor

Hi Mohammed,

many thanks for your support.

Attached the SIP Trunk Configuration and the SIP Profile.

T&R

Frank

Can you disable "Early Offer support for voice and video calls" and check if that makes any difference.

Regards,

Mohammed Noor

Hi Mohammed,

you are my hero.

After changing the settings like mentioned from you, it was working.

Many, many thanks for your great support!!!

Frank

I am glad I was able to help you.

Regards,

Mohammed Noor