Hello all,
Recently I added Linksys SPA-3102 for legacy phone line support here in the office. We have the following structure: there is a dedicated for 3CX Server software machine, version 7), there are few extensions on internal network (before the firewall) and there are few remote extensions which are in different parts of the city. At this moment we have good communication between all extensions (internal and external). After adding the SPA-3102 we got the access to our legacy phone line. It looks like it works well. At least when incomming call comes in it properly forward it to the Digital Receptionst, when we call outside using calling rule for this line it also works BUT:
Sometimes, very often, it disconnects the call by it self. For example when I call from my extension thru legacy phone line (SPA-3102) after some 15-20 seconds it just drop the line and in the lof file it says Call terminated. Or when someone calls to legacy line from outside and after that transfer to the extension, especcially if it is remote extension, it also may drop connection. Below I am attaching the log from such call when someone called from outside to the legacy line, got to teh digital receptionist and after that choose some remote extension (by remote I mean different city in this case and no tunnel, just SIP phone on other end connected through our firewall to our server).
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22:46:03.625 [CM503008]: Call(51): Call is terminated
22:46:03.609 [CM503008]: Call(51): Call is terminated
22:46:00.515 [CM503007]: Call(51): Device joined: sip:[email protected]:5060
22:46:00.515 [CM505001]: Ext.317: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA841-3.1.1(a)] Transport: [sip:192.168.0.51:5060]
22:46:00.515 [CM503002]: Call(51): Alerting sip:[email protected]:5060
22:45:57.515 [CM503024]: Call(51): Calling Ext:Ext.317@[Dev:sip:[email protected]:5060]
22:45:57.484 [CM503004]: Call(51): Route 1: Ext:Ext.317@[Dev:sip:[email protected]:5060]
22:45:57.484 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:45:52.765 [MS211000] C:51.1: 192.168.0.112:16414 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
22:45:40.468 [CM503007]: Call(51): Device joined: sip:[email protected]:40600;rinstance=bb1c9907d7f349fc
22:45:40.265 [CM503007]: Call(51): Device joined: sip:[email protected]:5062
22:45:40.265 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] Transport: [sip:127.0.0.1:5060]
22:45:40.265 [CM503002]: Call(51): Alerting sip:[email protected]:40600;rinstance=bb1c9907d7f349fc
22:45:38.984 [CM503024]: Call(51): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=bb1c9907d7f349fc]
22:45:38.984 [CM503004]: Call(51): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=bb1c9907d7f349fc]
22:45:38.984 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:45:38.984 [CM505002]: Gateway:[Videotron Gateway] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:192.168.0.51:5060]
22:45:38.593 [CM503001]: Call(51): Incoming call from 10000@(Ln.10000@Videotron Gateway) to <sip:[email protected]:5060>
22:45:38.593 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
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I know that this problem could be discussed already but I could not find the answer WHY this happens. Please help me if you can.
Thank you very much,
Regards,
Vlad
Recently I added Linksys SPA-3102 for legacy phone line support here in the office. We have the following structure: there is a dedicated for 3CX Server software machine, version 7), there are few extensions on internal network (before the firewall) and there are few remote extensions which are in different parts of the city. At this moment we have good communication between all extensions (internal and external). After adding the SPA-3102 we got the access to our legacy phone line. It looks like it works well. At least when incomming call comes in it properly forward it to the Digital Receptionst, when we call outside using calling rule for this line it also works BUT:
Sometimes, very often, it disconnects the call by it self. For example when I call from my extension thru legacy phone line (SPA-3102) after some 15-20 seconds it just drop the line and in the lof file it says Call terminated. Or when someone calls to legacy line from outside and after that transfer to the extension, especcially if it is remote extension, it also may drop connection. Below I am attaching the log from such call when someone called from outside to the legacy line, got to teh digital receptionist and after that choose some remote extension (by remote I mean different city in this case and no tunnel, just SIP phone on other end connected through our firewall to our server).
--------------------------------------------------------------------
22:46:03.625 [CM503008]: Call(51): Call is terminated
22:46:03.609 [CM503008]: Call(51): Call is terminated
22:46:00.515 [CM503007]: Call(51): Device joined: sip:[email protected]:5060
22:46:00.515 [CM505001]: Ext.317: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA841-3.1.1(a)] Transport: [sip:192.168.0.51:5060]
22:46:00.515 [CM503002]: Call(51): Alerting sip:[email protected]:5060
22:45:57.515 [CM503024]: Call(51): Calling Ext:Ext.317@[Dev:sip:[email protected]:5060]
22:45:57.484 [CM503004]: Call(51): Route 1: Ext:Ext.317@[Dev:sip:[email protected]:5060]
22:45:57.484 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:45:52.765 [MS211000] C:51.1: 192.168.0.112:16414 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
22:45:40.468 [CM503007]: Call(51): Device joined: sip:[email protected]:40600;rinstance=bb1c9907d7f349fc
22:45:40.265 [CM503007]: Call(51): Device joined: sip:[email protected]:5062
22:45:40.265 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] Transport: [sip:127.0.0.1:5060]
22:45:40.265 [CM503002]: Call(51): Alerting sip:[email protected]:40600;rinstance=bb1c9907d7f349fc
22:45:38.984 [CM503024]: Call(51): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=bb1c9907d7f349fc]
22:45:38.984 [CM503004]: Call(51): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=bb1c9907d7f349fc]
22:45:38.984 [CM503010]: Making route(s) to <sip:[email protected]:5060>
22:45:38.984 [CM505002]: Gateway:[Videotron Gateway] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:192.168.0.51:5060]
22:45:38.593 [CM503001]: Call(51): Incoming call from 10000@(Ln.10000@Videotron Gateway) to <sip:[email protected]:5060>
22:45:38.593 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
--------------------------------------------------------------------------------------------------------------------------------------
I know that this problem could be discussed already but I could not find the answer WHY this happens. Please help me if you can.
Thank you very much,
Regards,
Vlad
