'Internal' call issue with Hosted v10

Discussion in '3CX Phone System - General' started by purpleone, Jul 9, 2012.

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  1. purpleone

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    We are running 3cx version 10 Hosted Edition.
    The 3CX install in on a leased server located at a local data centre.
    We used Yealink T26 handsets at multiple client sites.
    The server currently has 5 instances of 3CX installed.
    We are ONLY having the following issue with one instance (one client site).

    Before I begin, my feeling is the issue is being caused by the firewall on the local router at the client site, but what is causing the issue is what I'm stuck on. The router being used is a Cisco SRP527. We have this router at other sites, with no issue.

    With the hosted version of 3cx ALL calls are considered external calls, as they come from outside the local network. The handsets are setup using STUN. The issue we are having is ONLY on calls between extensions. Inbound and outbound calls to mobiles, local numbers etc all work with out issue. When calls are made between 'local' extensions there is no audio. The calls go through ok, just no audio. We've even gone so far as to call in from an outside line, answer it (audio is ok), then perform an attended transfer to another extension (audio disappears), then when the call is released to the other extension the audio returns.

    I am seeing this, which I haven't seen before. It doesn't happen on every call and I'm not really sure what it means.

    17:13:06.268 Currently active calls - 1: [222]
    17:12:39.611 [CM503007]: Call(222): Device joined: sip:XXXXXX5575@sip.xxxxxx.com.au:5060
    17:12:39.609 [CM503007]: Call(222): Device joined: sip:200@XXclientipXX:5062
    17:12:39.604 [CM505003]: Provider:[PurpleOne] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sippy] PBX contact: [sip:XXXXXX5575@111.67.22.92:9060]
    17:12:39.604 [CM503002]: Call(222): Alerting sip:XXXXXX5575@sip.xxxxxx.com.au:5060
    17:12:36.232 Currently active calls - 1: [222]
    17:12:35.729 [MS205001] C:222.2 RTCP: Party address changed from 0.0.0.0:0 to XXX.XXX.XXX.XXX:45873
    17:12:33.930 [CM503025]: Call(222): Calling VoIPline:XXcallerXX@(Ln.10000@PurpleOne)@[Dev:sip:XXXXXX5575@sip.xxxxxx.com.au:5060]
    17:12:33.867 [CM503004]: Call(222): Route 1: VoIPline:XXcallerXX@(Ln.10000@PurpleOne)@[Dev:sip:XXXXXX5575@sip.xxxxxx.com.au:5060]
    17:12:33.866 [CM503010]: Making route(s) to <sip:XXcallerXX@XXhostipXX:9060>
    17:12:33.865 [CM505001]: Ext.200: Device info: Device Identified: [Man: Yealink;Mod: T28;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T28P 2.61.0.80] PBX contact: [sip:200@XXhostipXXX:9060]
    17:12:33.860 [CM503001]: Call(222): Incoming call from Ext.200 to <sip:XXcallerXX@XXhostipXX:9060>

    This is something I haven't come across before. So any ideas would be really helpful.
     
  2. wagner.damiao

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    Hello Nosha,

    can you make a test with sip proxy and discard the firewall.

    Follow the instructions:

    http://www.3cx.com/blog/releases/sip-proxy-manager/

    Best Regards.
     
  3. craigreilly

    craigreilly Well-Known Member

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    Have you looked at the local calling ports?
     
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  4. purpleone

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    Thanks for the reply guys. We had a 3cx training day here today and I was lucky enough to be able to ask Nicky about the problem in person. It was the local sip port and ftp ports. I didn't realise that each handset had to have unique ports. Changed them all - now working ok.
     
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