Incoming SIP call Fails after 1 ring when transfered from AA

[hassanalirazi]

Hi All,

I am facing an issue when the incoming call works when the DDI is sent directly to a phone but when we point to an AA and the call is transferred the call fails after 1 ring.

with a disconnect cause 41.

BYE sip:900861084181747@172.21.1.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.120.80.16:5060;branch=z9hG4bK8881129
From: <sip:00971529249845@10.120.80.16>;tag=821015D0-1B76
To: <sip:900861084181747@172.21.1.6>;tag=BF7169C0-26D9
Date: Mon, 27 May 2013 12:48:28 GMT
Call-ID: 85348CD5-C60211E2-949381F9-68B67898@10.120.80.16
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1369658912
CSeq: 103 BYE
Reason: Q.850;cause=41
P-RTP-Stat: PS=745,OS=13684,PR=1211,OR=24220,PL=0,JI=0,LA=0,DU=21
Content-Length: 0

May 27 12:48:33.086: //1863/736C6ADF09F4/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:900861084181747@172.21.1.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.120.80.16:5060;branch=z9hG4bK8881129
From: <sip:00971529249845@10.120.80.16>;tag=821015D0-1B76
To: <sip:900861084181747@172.21.1.6>;tag=BF7169C0-26D9
Date: Mon, 27 May 2013 12:48:28 GMT
Call-ID: 85348CD5-C60211E2-949381F9-68B67898@10.120.80.16
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1369658913
CSeq: 103 BYE
Reason: Q.850;cause=41
P-RTP-Stat: PS=745,OS=13684,PR=1211,OR=24220,PL=0,JI=0,LA=0,DU=21
Content-Length: 0

May 27 12:48:33.150: //1863/736C
GB-TPS-ROO-CR02#6ADF09F4/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.120.80.16:5060;branch=z9hG4bK8881129
From: <sip:00971529249845@10.120.80.16>;tag=821015D0-1B76
To: <sip:900861084181747@172.21.1.6>;tag=BF7169C0-26D9
Date: Mon, 27 May 2013 12:48:32 GMT
Call-ID: 85348CD5-C60211E2-949381F9-68B67898@10.120.80.16
Server: Cisco-SIPGateway/IOS-15.3.2.T
Timestamp: 1369658912
CSeq: 103 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=1313,OS=26260,PR=711,OR=13148,PL=0,JI=0,LA=0,DU=21
Content-Length: 0

Can someone please help me with this much appreciated.

Thanks

Hassan                  

[Ayodeji Okanlawon]

What products are you using? Ccme ? Cucm ? Gateways?
Describe your call flow..

Also please attach the full debug. Include calling and called number.


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[hassanalirazi]

Hi

The call flow is SIP TRUNK  <---------> SIP-GW <---- H323----> CE-Router <------SIP----> Customer GW <---- SIP ----> CUCM.

The phone rings once and then a BYE is sent from our CE Router to the Customer GW.

Thanks

Hassan

[Ayodeji Okanlawon]

I suggest you upgrade the IOS on the CUBE gateway GB-TPS-ROO-CR02. The sip traces on the gateway have so many repetition on it. I am not sure if this is a defect or its the way you have obtained the logs. It makes it difficult for me to correlate things. Upgrade the IOS to a 15.X series eg 15.1M4 and test again

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[hassanalirazi]

Thanks i will hopefully be able to upgrade the IOS. Can you please tell me if there already is a defect that would cause repetition?

The gateway has Version 15.1(4)M4 already.

Also i get internal server error when i call from a PSTN circuit the rest of the call flow remains the same.

Thanks

[Ayodeji Okanlawon]

Oh ok. Then that IOS should be fine..Its one of the stable ones. Did you have debug ccsip all turned on? Can you just turn on debug ccsip messages...and do a test again..send the logs

What do you mean you get internal server error when you call from a PSTN circuit? can you explain better

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[hassanalirazi]

We have incoming SIP DDIS and Simple PSTN DDIs when i call from the SIP one the call always disconnects giving a cause 41. When i dial from PSTN the call connects at times and at times it gives an Internal Error.

I am attaching the logs of the call that always fails over SIP

[Ayodeji Okanlawon]

What is the calling and called number..this trace doesnt look like a straigh forward call..Looks like it goes to voicemail..is this a call to auto attendant?

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[hassanalirazi]

yes the call goes to the AA the call connects fine .... when i enter the extension the called phone rings once and the call fails.

The calling Number is 0097152924984 called number is 00861084181747

[hassanalirazi]

Still facing this issue guys .....