SIP Transfer on CME

[Christian Semmler]

Hello,

I'm not able to transfer a call from a SIP phone to another phone.For testing I use the Eyebeam Client, but I recognized the problem with other SIP devices too(SPAxxx).

I set up a light configuration in the lab with two SCCP phones and one SIP phone. Trying to transfer a call from the SCCP->SIP phone to the other SCCP phone. But everytime I'm getting a busy signal and all calls are dropped.

If I initiate the two calls from the SIP phone and connect the this two calls everything is fine, conferencing with all three, too.

voice service voip

allow-connections sip to sip

sip

registrar server expires max 1200 min 300

!

!

voice register global

mode cme

source-address 10.0.16.1 port 5060

max-dn 56

max-pool 14

authenticate register

tftp-path flash:

create profile sync 0000315004002712

!

voice register dn  2

number 35

name TEST-PC

label TEST-PC

!

voice register pool  3

id mac F0DE.F1D1.CBB5

number 1 dn 2

dtmf-relay sip-notify

username pc_test password test_pc

codec g711ulaw

best regards

Christian

[Ayodeji Okanlawon]

Christian,

You might need an MTP for a succesful transfer between sccp and sip phone..

Can you send

debug ccsip messages

debug h225 asn1

debug h245 asn1

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[Christian Semmler]

Hi,

thanks for your answer. I tried it today with MTP but the problem still occur.

dspfarm profile 3 mtp

codec g711ulaw

codec pass-through

maximum sessions hardware 4

associate application SCCP

TGZ_UC500#sh dspfa prof 3

Dspfarm Profile Configuration

Profile ID = 3, Service = MTP, Resource ID = 3

Profile Description :

Profile Service Mode : Non Secure

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP   Status : ASSOCIATED

Resource Provider : FLEX_DSPRM   Status : UP

Number of Resource Configured : 4

Number of Resource Available : 4

Hardware Configured Resources : 4

Hardware Available Resources : 4

Software Resources : 0

Codec Configuration: num_of_codecs:2

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : pass-through, Maximum Packetization Period : 0

I attached you the requested logs.

regards

Christian

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[Ayodeji Okanlawon]

Hi I have looked at the logs and the call seem to fail at the NOTIFY for the sip refer...is extension 10 the sccp phone..

Do you have transfer-pattern .T configured under telephony-service?

does transfer work between sip to sip phone..can you do a test and send debug ccsip messages for sip to sip trasfer

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[Christian Semmler]

Hi,

transfer-pattern .T and full-consult is configured.

SIP to SIP transfer isn't working, too. I attached you the debug + a debug of a working blind transfer and the working transfer if both calls are initiated from the SIP phone.

regards

Christian

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[Ayodeji Okanlawon]

Please send your full sh run

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[Christian Semmler]

Hi,

attached the sh run file.     

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[Ayodeji Okanlawon]

Christian,

Looking at the l ogs I have a few recommendations..

1. Your IP phones are in the subnet 192.168.15.0...I suggest you configure your ccme source address to 192.168.15.1 not 10.0.0.1..This makes trouble shooting easy

Attached is the flow graph of your sip-ladder...Its quite difficult to troubleshoot the way you have things setup...

So please make the change..

2. Here is the ananlysis of the sip to sip call logs..

1. Extension 35(192.168.15.100) calls extension 34..(192.168.15.4)

2. extension 34 puts 35 on hold, sends a re-invite to ext 34

3. Extension 34 places a new call to extension 36 (192.168.15.6)

4. extension 34 puts 36 on hold (attempt ot transfer) so we get a re-nvite to 36
5. we get a refer extension 36 to 35 by 34  (Refer for the transfer which extension 34 is doing)

6. finally we try to connect extenion 35 to 36..we send invite to 35 from 36...Then gateway sends a moved temporarily because xtension 35 is forwarded to extension 34

7. Next gateway send invite extension 34...From there the call fails....With 487 call doesnt exist and 500 Internal server error

The call is getting into a loop because you have forwarded extension 35 to extension 34....Please remove the call forward and try again...

If that doesnt work for the sip to sip transfer please configure the ff and test again...

3 configure this on your gateway

voice service voip
no supplementary-service sip refer
no supplementary-service sip moved-temporarily

NB: Take the debug ccsip messages for the sip to sip transfer after you removed the call forwarding

Also send a seperate debug ccsip messages when you have made the configuration changes as speficied in step 3 above

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[Christian Semmler]

Hi,

thanks for your intensive support.

There is no forward configured, I installed alle SIP Clients brand new just for the test in my lab.

I attached the two debugs, this time with comments what I do.

I changed the IP of the SIP and SCCP Callmanager. There was no change after I configured no supplementary-service.

debug_SIP_comments.txt = debug with comments.

debug_no_supp.txt = no supplementary-service sip refer and move-temp

regards

Christian

[Ayodeji Okanlawon]

OK..give me some time to look at this..I will comeback to you later

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[Christian Semmler]

Hi,

do you found something in the logs?

regards

Christian

[Ayodeji Okanlawon]

Christian,

Can you change your cme source address to that of 192.168.15.1 instead of the 10.0.0.1...Please do that and test again..Send dbeug ccsip  meesages

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[Ayodeji Okanlawon]

Looking at the old logs, the problem is that the system sees Extension 35 is Callforwarded t Extension 34..

++=Here we see when the trasnfer is about to be cmpleted, the gateway sends invite to 35 from 36+++

Received:

INVITE sip:35@192.168.15.1 SIP/2.0

Via: SIP/2.0/UDP 192.168.15.6:31924;branch=z9hG4bK-d87543-4d034b371a74ea1c-1--d87543-;rport

Max-Forwards: 70

Contact: <36>

To: <35>

From: "pc3"<36>;tag=f47ee23f

Call-ID: 676307120a3d301dNDU3MTEzMmRkOTI2NTMyNzNiZWJlZTk0NWE1NWU3ZGU.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1003s stamp 31159

Referred-By: <34>

Replaces: df711e40cb11df0fZjdjYzIxY2M4ODQ2ZGIxNTU2NWEyOGJmZmQwNjRlZjY.;to-tag=9BED7E8-CC2;from-tag=205dbd37

Content-Length: 441

Apr 24 15:15:26.880: //586/A0AA510A832E/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.15.6:31924;branch=z9hG4bK-d87543-4d034b371a74ea1c-1--d87543-;rport

From: "pc3"<36>;tag=f47ee23f

To: <35>

Date: Wed, 24 Apr 2013 15:15:26 GMT

Call-ID: 676307120a3d301dNDU3MTEzMmRkOTI2NTMyNzNiZWJlZTk0NWE1NWU3ZGU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Apr 24 15:15:26.888: //586/A0AA510A832E/SIP/Msg/ccsipDisplayMsg:

+++Here the gateways sees extension 35 is forwarded t extenion 34+++

Sent:

SIP/2.0 302 Moved Temporarily

Via: SIP/2.0/UDP 192.168.15.6:31924;branch=z9hG4bK-d87543-4d034b371a74ea1c-1--d87543-;rport

From: "pc3"<36>;tag=f47ee23f

To: <35>;tag=9BF7B98-1897

Date: Wed, 24 Apr 2013 15:15:26 GMT

Call-ID: 676307120a3d301dNDU3MTEzMmRkOTI2NTMyNzNiZWJlZTk0NWE1NWU3ZGU.

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Diversion: <35>;reason=unconditional;counter=1

Contact: <34>

Content-Length: 0

+++Next the gateway tries to send an IvITE to extension 34, which is where the call fails..because we almost get into a loop+++

Received:

INVITE sip:34@192.168.15.1 SIP/2.0

Via: SIP/2.0/UDP 192.168.15.6:31924;branch=z9hG4bK-d87543-d808710dc94dde3f-1--d87543-;rport

Max-Forwards: 70

Contact: <36>

To: <35>

From: "pc3"<36>;tag=f47ee23f

Call-ID: 676307120a3d301dNDU3MTEzMmRkOTI2NTMyNzNiZWJlZTk0NWE1NWU3ZGU.

CSeq: 2 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: eyeBeam release 1003s stamp 31159

Referred-By: <34>

Replaces: df711e40cb11df0fZjdjYzIxY2M4ODQ2ZGIxNTU2NWEyOGJmZmQwNjRlZjY.;to-tag=9BED7E8-CC2;from-tag=205dbd37

Content-Length: 441

Have yu restarted the gateway at all...If you are sure there is no call forwarding try and restart the gateway..Did you configure callf orward all on extension 35 at any time?

[Christian Semmler]

Hi,

thanks for your lookup.

I changed the IP to 192.168.15.1 and changed the domain on the SIP Clients to this adress, too.

I checked for call-forwarding, but there is no configured.

Attached a new debug.

1st Call from 35 to 34

2nd Call from 34 to 36

then Transfer on 34.

regards

Christian

[Ayodeji Okanlawon]

Christian,

I have had a look and I dont see the "moved temporarily messages" again..Have you done the transfer without outting the secnd call on hold

so 35 calls 34, 34 puts 35 on hold

34 then calls 36, instead of putting 36 on hold, just hit the transfer button and see if that works...

If it doesnt then you may have to contact Cisco..I am not sure why it wont work

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