cannot do an attended call transfer

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rheijmans

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just noticed a problem with my fresh 3cx installation for 60 users.

when doing an attended call xfer (external call coming in) from 1 3CX extension (501) to another (502) the attended call xfer works but only has 1 way audio. the called party cannot here the original external party that got transfered. when doing a blind xfer it works good?!?

blind xfer:
external party calls --> 501 --> press xfer button and type 502 and press xfer button again --> call is directly being transferred to 502 --> 502 picks up and there is good audio both ways

attended xfer:
external party calls --> 501 --> press xfer button and type in 502 and press Ok --> 502' picks up and here's 501 --> 501 says i have a call for you and presses xfer button again --> 502 now has the external caller on the line but cannot hear him. The external party can hear 502 well.

i have tried several things but none work. not sure why this is. i tried turning off/on PBX deliver audio, turn off reinvite/replace options on gateway or in extension properties, nothing worked yet.

has anyone an idea what might be the problem?

thx

Ray
 
Hi Ray,

http://www.3cx.com/forums/forum-rules-read-to-get-answers-93.html

...
2. List what SIP phones and VOIP provider or VOIP Gateway you are using
3. POST THE SERVER STATUS LOG (This can be found in the management console - Phone System > Server status) Clear the status log, then perform the action that causes a problem, and then post that part of the log with your post.

Could you please describe scenario of the problematic call and specify partcipants(devices)?

Thanks
 
hi sy,

sorry for the incomplete port. i will fill in the missing parts,
3CX v9 SP 5
sip phones (ext. 501 and 502) = tiptel IP 284
voip gateway = cisco 2811
external caller = could be any phone (in this example i used an older sopho pbx phone (isdn) with caller id 31541589801

Problem description:
external caller (31541589801) calls 3CX ext. 501 via gateway cisco2811. ext. 501 answers and caller asks to be transferred to ext. 502. 501 does an attended transfer by pressing xfer and 502 then Ok button. problem is that 31541589801 and ext 502 have 1 way audio. ext 502 cannot hear 31541589801 but 31541589801 CAN here ext. 502.

the weird thing is that a blind xfer is going ok.

call flow: 31541589801 --> cisco2811 gateway --> ext. 501 --> xfering call to 502


3CX server log:


Please remember to used code tags when posting logs - MFM
Code:
22:18:50.124  [MS210003] C:45.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.206.21:7240(7241)
22:18:50.124  [MS210000] C:45.1:Offer received. RTP connection: 192.168.206.3:17070(17071)
22:18:50.124  Remote SDP is set for legC:45.1
22:18:49.968  [CM503008]: Call(46): Call is terminated
22:18:49.953  Replaces: C:46.2
22:18:49.953  Refer: from=<sip:[email protected]>;tag=1901884657; to="31541589801"<sip:[email protected]:5060>;tag=a0027a13; RefTo=<sip:[email protected]?Replaces=1701566337%3Bto-tag%3D810e666f%3Bfrom-tag%3D739300433>
22:18:45.812  Currently active calls - 2: [45,46]
22:18:44.359  Session 4065 of leg C:46.1 is confirmed
22:18:44.015  [CM503007]: Call(46): Device joined: sip:[email protected]:5062
22:18:44.015  [CM503007]: Call(46): Device joined: sip:[email protected]:5062
22:18:44.015  [MS210003] C:46.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.206.21:7246(7247)
22:18:43.999  [MS210001] C:46.2:Answer received. RTP connection[unsecure]: 10.30.4.159:11782(11783)
22:18:43.999  Remote SDP is set for legC:46.2
22:18:42.437  [CM505001]: Ext.502: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Tiptel IP 284 6.51.13.1 00156517b73d] PBX contact: [sip:[email protected]:5060]
22:18:42.437  [CM503002]: Call(46): Alerting sip:[email protected]:5062
22:18:41.827  [CM503025]: Call(46): Calling Ext:Ext.502@[Dev:sip:[email protected]:5062]
22:18:41.827  [MS210002] C:46.2:Offer provided. Connection(transcoding mode): 192.168.206.21:7248(7249)
22:18:41.780  [CM503004]: Call(46): Route 1: Ext:Ext.502@[Dev:sip:[email protected]:5062]
22:18:41.780  [CM503010]: Making route(s) to <sip:[email protected]>
22:18:41.780  [MS210000] C:46.1:Offer received. RTP connection: 10.30.4.193:11794(11795)
22:18:41.780  Remote SDP is set for legC:46.1
22:18:41.780  [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Tiptel IP 286 2.51.13.7 00156515332c] PBX contact: [sip:[email protected]:5060]
22:18:41.780  [CM503001]: Call(46): Incoming call from Ext.501 to <sip:[email protected]>
22:18:41.780  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/UDP 10.30.4.193:5062;rport=5062;branch=z9hG4bK488713953
  Max-Forwards: 70
  Contact: <sip:[email protected]:5062>
  To: <sip:[email protected]>
  From: "Raymond Heijmans"<sip:[email protected]>;tag=739300433
  Call-ID: 1701566337
  CSeq: 2 INVITE
  Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  Proxy-Authorization: Digest username="501",realm="3CXPhoneSystem",nonce="414d535c035120b108:0105d3469c1f09496b1d7ae55e606b73",uri="sip:[email protected]",response="35b97d6bfe1445fbba7c1e3b584219fd",algorithm=MD5
  Supported: replaces
  User-Agent: Tiptel IP 286 2.51.13.7 00156515332c
  Allow-Events: talk, hold, conference, refer, check-sync
  Content-Length: 0
  
22:18:34.936  [MS210003] C:45.3:Answer provided. Connection(transcoding mode[unsecure]):192.168.206.21:7244(7245)
22:18:34.936  [MS210001] C:45.1:Answer received. RTP connection[unsecure]: 192.168.206.3:17070(17071)
22:18:34.936  Remote SDP is set for legC:45.1
22:18:34.811  [MS210002] C:45.1:Offer provided. Connection(transcoding mode): 192.168.206.21:7240(7241)
22:18:34.811  [MS210000] C:45.3:Offer received. RTP connection: 10.30.4.193:11792(11793)
22:18:34.811  Remote SDP is set for legC:45.3
22:18:31.264  [CM503003]: Call(45): Call to sip:[email protected]:5060 has failed; Cause: 487 Request Terminated; from IP:192.168.206.101:4002
22:18:31.171  Session 4042 of leg C:45.1 is confirmed
22:18:31.046  [CM503007]: Call(45): Device joined: sip:[email protected]:5062
22:18:31.046  [CM503007]: Call(45): Device joined: sip:[email protected]
22:18:31.046  [MS210003] C:45.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.206.21:7240(7241)
22:18:31.046  [MS210001] C:45.3:Answer received. RTP connection[unsecure]: 10.30.4.193:11792(11793)
22:18:31.046  Remote SDP is set for legC:45.3
22:18:30.499  [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Tiptel IP 286 2.51.13.7 00156515332c] PBX contact: [sip:[email protected]:5060]
22:18:30.499  [CM503002]: Call(45): Alerting sip:[email protected]:5062
22:18:29.577  [CM505001]: Ext.501: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 5.0.14900.0] PBX contact: [sip:[email protected]:5060]
22:18:29.577  [CM503002]: Call(45): Alerting sip:[email protected]:4002;rinstance=e2af4eed2bd8f51b
22:18:29.452  [CM503025]: Call(45): Calling Ext:Ext.501@[Dev:sip:[email protected]:5062]
22:18:29.452  [MS210002] C:45.3:Offer provided. Connection(transcoding mode): 192.168.206.21:7244(7245)
22:18:29.436  [CM503025]: Call(45): Calling Ext:Ext.501@[Dev:sip:[email protected]:4002;rinstance=e2af4eed2bd8f51b]
22:18:29.436  [MS210002] C:45.2:Offer provided. Connection(transcoding mode): 192.168.206.21:7242(7243)
22:18:29.389  [CM503004]: Call(45): Route 1: Ext:Ext.501@[Dev:sip:[email protected]:4002;rinstance=e2af4eed2bd8f51b,Dev:sip:[email protected]:5062]
22:18:29.389  [CM503010]: Making route(s) to <sip:[email protected]>
22:18:29.389  [MS210000] C:45.1:Offer received. RTP connection: 192.168.206.3:17070(17071)
22:18:29.389  Remote SDP is set for legC:45.1
22:18:29.389  [CM505001]: Ext.31541589801: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-SIPGateway/IOS-12.x] PBX contact: [sip:[email protected]:5060]
22:18:29.389  [CM503001]: Call(45): Incoming call from Sip.31541589801 to <sip:[email protected]>
22:18:29.389  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected]:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.206.3:5060;branch=z9hG4bK1B77AE
  Max-Forwards: 68
  Contact: <sip:[email protected]:5060>
  To: <sip:[email protected]>
  From: "31541589801"<sip:[email protected]>;tag=F9D008-710
  Call-ID: [email protected]
  CSeq: 101 INVITE
  Expires: 180
  Min-SE: 1800
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
  Date: Wed, 26 Jan 2011 21:28:03 GMT
  Supported: 100rel, timer, resource-priority, replaces, sdp-anat
  Timestamp: 1296077283
  User-Agent: Cisco-SIPGateway/IOS-12.x
  Allow-Events: telephone-event
  Content-Length: 0
  Remote-Party-ID: "31541589801" <sip:[email protected]>;party=calling;screen=no;privacy=off
  Cisco-Guid: 4258359513-0684265952-2188148914-1804660194

Hope this clears a bit..


thx

Ray
 
another thing that i just noticed is that when ext. 501 does the attended xfer to ext. 502. and 502 is connected to the external call +31541589801 and presses "Hold" and "Resume" on the tiptel phone...the audio suddenly works both ways ?!?!?
 
For the devices which are not officially supported (tiptel for example)
(recommended)
Compatibility mode:
"Supports re-Invites=off"

(to try)
Advanced mode. It allows to avoid duplication of traffic in local network. Often it works:
"Supports re-Invites=on"
"Support replaces=off"
"PBX delivers audio=off"

(try it if you are sure that there are no any problems in "Advanced mode")
Full accelerated(default configuration):
"Supports re-Invites=on"
"Support replaces=on"
"PBX delivers audio=off"

P.S. details are described in http://www.3cx.com/forums/forum-rules-read-to-get-answers-93.html
 
Stepan,

do you mean to change this per extension or in the 2811 gateway properties?

right now i have following set for all extension properties

"Supports re-Invites=off"
"Support replaces=off"
"PBX delivers audio=off"

and for the 2811 gateway i have:

"Supports re-Invites=on"
"Support replaces=on"
"PBX delivers audio=on"
 
Ray,

All aspects of our conversation are specified in the post http://www.3cx.com/forums/forum-rules-read-to-get-answers-93.html

Unfortunately, it is really difficult to devine what is going wrong :(

Thank you for your understanding

P.S. I meant only things which are specified in my post. If you are using unsupported equipment then you need to start with "compatibility mode"... More details=More help from community.
 
Hi Stepan,

I noticed that in the latest update for 3CX is support for tiptel provisioning. Why isn't tiptel 28x then on the list of compatible hardware?

thx

Ray
 
hi Stepan,

just to let you know that i have tested with the 3CX softphone client and could reproduce the same problem. it has nothing to do with a compatible phone or not. somehow 3CX doesn't correctly do an attended transfer. i will test the options you send me although i already did a lot of testing before but just to make sure i did not miss anything.

ray
 
turning off reinvite/replace in the gateway (cisco 2811) seems to do the trick. i get 2 way audio again when doing an attended transfer....but in another situation mobile nr. --> ext. 501 --> external number..... again 1 way audio.

i think that the cisco2811 does not handle the reinvite/replace correctly in certian attended transfer situations.

putting in a sip proxy between the 3CX and cisco 2811 (3CX <--> SIP Proxy <--> Cisco 2811) seems to work ok for everything.... all attended transfer situations work. not sure why then the 3CX <--> Cisco has trouble with reinvite/replace when communicating directly with each other. time to dig up cisco schoolbooks again.
 
Hi Ray,

Thanks a lot for update. I'm glad to hear that the workaround was found. Probably, it was enough to setup connection with Cisco as "Support replaces=off". Sorry, but I wasn't able to divine it basing on the information provided in this topic :(
 
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