04-21-2013 07:27 AM - edited 03-16-2019 04:55 PM
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
04-21-2013 12:57 PM
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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04-22-2013 11:12 AM
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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04-22-2013 03:56 PM
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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04-23-2013 01:36 PM
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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04-23-2013 02:05 PM
Please send your full sh run
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
04-23-2013 02:17 PM
04-24-2013 05:27 AM
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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04-24-2013 08:32 AM
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
04-24-2013 08:47 AM
OK..give me some time to look at this..I will comeback to you later
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04-30-2013 01:24 AM
Hi,
do you found something in the logs?
regards
Christian
04-30-2013 01:39 AM
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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04-30-2013 02:14 AM
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>36>
To: <35>35>
From: "pc3"<36>;tag=f47ee23f36>
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>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=f47ee23f36>
To: <35>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=f47ee23f36>
To: <35>;tag=9BF7B98-189735>
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=135>
Contact: <34>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>36>
To: <35>35>
From: "pc3"<36>;tag=f47ee23f36>
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>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?
05-01-2013 11:39 AM
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
05-01-2013 12:42 PM
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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"opportunity is a haughty goddess who waste no time with those who are unprepared"
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