Transferring when using Cloud 3CX

Discussion in '3CX Phone System - General' started by kristijonas, Apr 22, 2015.

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  1. kristijonas

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    Hi all,
    trying to figure out problems with call transfers.
    When i transfer the call, it still shows as "transferring" even though both parties can already talk. Can only end it by clicking "end call". Both parties are still able to talk each other though..
    Also i get strange/interesting messages in the logs:

    Let's say (numbers are changed) 9999999999 called 1111111111. Then 103 picked up the call and transferred to 105.

    222.222.222.222 is our public ip,
    74.112.200.122 - cloud pbx's ip,
    67.102.144.50 - sip trunks ip


    22-Apr-2015 12:13:41.209 Leg L:2164.1[Line:10001<<9999999999] is terminated: Cause: BYE from PBX
    22-Apr-2015 12:13:41.209 [CM503008]: Call(C:2164): Call is terminated
    22-Apr-2015 12:13:41.208 Leg L:2164.3[Extn] is terminated: Cause: BYE from 222.222.222.222:12163
    22-Apr-2015 12:13:41.005 Leg L:2164.4[Extn] is terminated: Cause: BYE from 222.222.222.222:12165
    22-Apr-2015 12:13:21.915 Currently active calls - 1: [2164]
    22-Apr-2015 12:13:20.930 [MS105000] C:2309.1: No RTP packets were received:remoteAddr=74.112.200.122:15730,extAddr=0.0.0.0:0,localAddr=74.112.200.122:58628
    22-Apr-2015 12:12:59.475 [MS104000] C:2164.1: (ACTIVE) RTP stream to 67.102.144.50:51412 restored.
    22-Apr-2015 12:12:59.240 Leg L:2165.1[Extn] is terminated: Cause: BYE from PBX
    22-Apr-2015 12:12:59.210 [CM503008]: Call(C:2165): Call is terminated
    22-Apr-2015 12:12:58.021 [CM503007]: Call(C:2165): Extn:105 has joined, contact <sip:105@10.0.1.105:12165>
    22-Apr-2015 12:12:58.020 [CM503007]: Call(C:2165): Extn:103 has joined, contact <sip:103@10.0.1.103:12163>
    22-Apr-2015 12:12:58.019 L:2165.2[Extn] has joined to L:2165.1[Extn]
    22-Apr-2015 12:12:56.289 [CM503025]: Call(C:2165): Calling T:Extn:105@[Dev:sip:105@10.0.1.105:12165] for L:2165.1[Extn]
    22-Apr-2015 12:12:56.240 [CM503027]: Call(C:2165): From: Extn:103 ("Marry" <sip:103@74.112.200.122:5060>) to T:Extn:105@[Dev:sip:105@10.0.1.105:12165]
    22-Apr-2015 12:12:56.240 [CM503004]: Call(C:2165): Route 1: from L:2165.1[Extn] to T:Extn:105@[Dev:sip:105@10.0.1.105:12165]
    22-Apr-2015 12:12:56.239 [CM503001]: Call(C:2165): Incoming call from Extn:103 to <sip:105@74.112.200.122:5060>
    22-Apr-2015 12:12:55.354 [MS004000] C:2164.1: (ACTIVE): Can't send RTP stream to 0.0.0.0:51412 destination unreachable
    22-Apr-2015 12:12:53.154 [MS104000] C:2164.1: (ACTIVE) RTP stream to 67.102.144.50:51412 restored.
    22-Apr-2015 12:12:53.126 Leg L:2164.2[Ivr] is terminated: Cause: BYE from 127.0.0.1:5483
    22-Apr-2015 12:12:52.922 [CM503007]: Call(C:2164): Extn:103 has joined, contact <sip:103@10.0.1.103:12163>
    22-Apr-2015 12:12:52.920 L:2164.3[Extn] has joined to L:2164.1[Line:10001<<9999999999]
    22-Apr-2015 12:12:50.605 [CM503025]: Call(C:2164): Calling T:Extn:103@[Dev:sip:103@10.0.1.103:12163] for L:2164.1[Line:10001<<9179031912]
    22-Apr-2015 12:12:50.598 [CM503027]: Call(C:2164): From: Line:10001<<9999999999 ("Bob" <sip:9179031912@mysipdomain.local:5060>) to T:Extn:103@[Dev:sip:103@10.0.1.103:12163]
    22-Apr-2015 12:12:50.598 [CM503004]: Call(C:2164): Route 1: from L:2164.1[Line:10001<<9999999999] to T:Extn:103@[Dev:sip:103@10.0.1.103:12163]
    22-Apr-2015 12:12:50.454 [MS004000] C:2164.1: (ACTIVE): Can't send RTP stream to 0.0.0.0:51412 destination unreachable
    22-Apr-2015 12:12:49.913 Currently active calls - 1: [2164]
    22-Apr-2015 12:12:41.482 [CM503007]: Call(C:2164): Ivr:900 has joined, contact <sip:900@127.0.0.1:5483>
    22-Apr-2015 12:12:41.481 [CM503007]: Call(C:2164): Line:10001<<9999999999 has joined, contact <sip:1111111111@67.102.144.50:5060>
    22-Apr-2015 12:12:41.480 L:2164.2[Ivr] has joined to L:2164.1[Line:10001<<9999999999]
    22-Apr-2015 12:12:41.330 [CM503025]: Call(C:2164): Calling T:Ivr:900@[Dev:sip:900@127.0.0.1:5483;rinstance=db3c5ef0da557aa3] for L:2164.1[Line:10001<<9999999999]
    22-Apr-2015 12:12:41.284 [CM503027]: Call(C:2164): From: Line:10001<<9999999999 ("Bob" <sip:9999999999@mysipdomain.local:5060>) to T:Ivr:900@[Dev:sip:900@127.0.0.1:5483;rinstance=db3c5ef0da557aa3]
    22-Apr-2015 12:12:41.284 [CM503004]: Call(C:2164): Route 1: from L:2164.1[Line:10001<<9999999999] to T:Ivr:900@[Dev:sip:900@127.0.0.1:5483;rinstance=db3c5ef0da557aa3]
    22-Apr-2015 12:12:41.284 [CM505003]: Provider:[megapath] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:1111111111@74.112.200.122:5060]
    22-Apr-2015 12:12:41.283 [CM503001]: Call(C:2164): Incoming call from Line:10001<<9999999999 to <sip:900@74.112.200.122:5060>

    -------------------------------------------------

    Phones are provisioned using STUN, TCP and UDP ports are port-forwarded and allowed on the firewall.
    Also when i click "phones" in the web gui, it shows private ip's for "IP of Phone", not the public ones. Is that the expected behavior?
    Tried my luck with "supports re-invite" and "supports replace" - no luck. Do i need to restart some 3cx services after i do such changes to my SIP provider's config?

    Not using SBCs or proxies in any of our offices (want to remove as many single points of failure as possible).

    Thanks for any thoughts!
     
  2. millennium2

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    How many phones at the remote office?

    I would use a SBC Controller and be done with it. I prefer Raspberry Pi over a Windows PC. Stable, reliable, and incredibly cheap so its easily replaceable if it ever fails on you.
     
  3. kristijonas

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    Each office has a couple of phones.
    I guess will have to go with using SBC. Just tried it and transferring works fine.
    One issue though. When selecting Cisco 504G - i can pick 3CX SBC as a provisioning method.
    When using Cisco 504G + 500S - there's no such method listed in the drop down menu. Is there a real difference between SBC and SIP Proxy Manager?
     
  4. kristijonas

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    Compared both template files, 504G+500S is an older version, does not have SBC config option and some details. Will have to merge them.
     
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