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Callers hang up, but calls remain & phones are unavailable

Discussion in '3CX Phone System - General' started by RobertKroll, Sep 1, 2009.

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  1. RobertKroll

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    For the most part the system has been working flawlessly, however (you knew that was coming didn't ya)...For some strange reason we are having one tremendous issue that may lead us in taking the phone system down. I am running the newest release of the system along with a Patton 4 port and SNOM telephones. We are running on a Microsoft server 2008 box with NOTHING else on it. What is happening is that when somebody dials any number (including voicemail...999), or receives an incoming call, the call never disconnects even after both parties have disconnected. I have to log into the manager and manually disconnect the calls. The major problem with this is that we are getting floods of messages when people are sitting at their desk doing nothing. During the day, we route calls directly to handsets instead of Automated Attendant. We have tried just about everything, and I have trolled through the community, but I haven't been able to come up with a solution. I am sure it is one of the thousands of little settings that we have overlooked...

    If we can get past this, we are also going to need a professional to guide us through some other setup issues that are nowhere near as pressing as the above problem.

    Regards,
    Bob
     
  2. leejor

    leejor Well-Known Member

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    A post of the 3CX system log, showing a call from beginning to end (even though it doesn't disconnect) would be helpfull.
     
  3. RobertKroll

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    The following is the whole log as it exists. Please try to ignore the remote telephone that won't work properly. I have been working on that one for months, but it is not too important right now.


    17:08:47.648 [CM504001]: Ext.101: new contact is registered. Contact(s): [sip:101@192.168.1.216:2051;line=z68xfxs0/101]

    17:08:13.647 [CM503008]: Call(1469): Call is terminated

    17:08:13.647 [CM502001]: Source info: From: "501"<sip:501@office.temp-art.net:5060>;tag=6e50db8188"100"<sip:100@office.temp-art.net:5060>

    17:08:13.647 [CM503013]: Call(1469): Incoming call rejected, caller is unknown; msg=SipReq: INVITE 100@office.temp-art.net:5060 tid=f8eb103a60d972dce cseq=INVITE contact=501@192.168.1.200:5060 / 5811 from(wire)

    17:08:13.437 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

    INVITE sip:100@office.temp-art.net:5060 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK4f690bd83f91cfd1e;received=173.56.44.206

    Max-Forwards: 70

    Contact: "501"<sip:501@192.168.1.200:5060;transport=udp>

    To: "100"<sip:100@office.temp-art.net:5060>

    From: "501"<sip:501@office.temp-art.net:5060>;tag=6e50db8188

    Call-ID: 73dad46fc82cffd3

    CSeq: 5810 INVITE

    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO

    Supported: timer, 100rel, replaces

    User-Agent: Aastra 53i/2.2.0.166

    Allow-Events: talk, hold, conference

    Content-Length: 0




    17:08:13.437 [CM302001]: Authorization system can not identify source of: SipReq: INVITE 100@office.temp-art.net:5060 tid=4f690bd83f91cfd1e cseq=INVITE contact=501@192.168.1.200:5060 / 5810 from(wire)

    17:08:06.630 [CM503008]: Call(1468): Call is terminated

    17:08:06.630 [CM502001]: Source info: From: "501"<sip:501@office.temp-art.net:5060>;tag=112377ab0b"100"<sip:100@office.temp-art.net:5060>

    17:08:06.630 [CM503013]: Call(1468): Incoming call rejected, caller is unknown; msg=SipReq: INVITE 100@office.temp-art.net:5060 tid=3c556f8190a6d4a3f cseq=INVITE contact=501@192.168.1.200:5060 / 10415 from(wire)

    17:08:06.419 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

    INVITE sip:100@office.temp-art.net:5060 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK229b491b2896fcc00;received=173.56.44.206

    Max-Forwards: 70

    Contact: "501"<sip:501@192.168.1.200:5060;transport=udp>

    To: "100"<sip:100@office.temp-art.net:5060>

    From: "501"<sip:501@office.temp-art.net:5060>;tag=112377ab0b

    Call-ID: 9fbde56f871e5dd1

    CSeq: 10414 INVITE

    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO

    Supported: timer, 100rel, replaces

    User-Agent: Aastra 53i/2.2.0.166

    Allow-Events: talk, hold, conference

    Content-Length: 0




    17:08:06.419 [CM302001]: Authorization system can not identify source of: SipReq: INVITE 100@office.temp-art.net:5060 tid=229b491b2896fcc00 cseq=INVITE contact=501@192.168.1.200:5060 / 10414 from(wire)

    17:07:55.690 [CM503008]: Call(1467): Call is terminated

    17:07:55.690 [CM502001]: Source info: From: "501"<sip:501@office.temp-art.net:5060>;tag=03665448cf"100"<sip:100@office.temp-art.net:5060>

    17:07:55.690 [CM503013]: Call(1467): Incoming call rejected, caller is unknown; msg=SipReq: INVITE 100@office.temp-art.net:5060 tid=6c83ea7b018c63ba2 cseq=INVITE contact=501@192.168.1.200:5060 / 26242 from(wire)

    17:07:55.465 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:

    INVITE sip:100@office.temp-art.net:5060 SIP/2.0

    Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKc097d65f23a380942;received=173.56.44.206

    Max-Forwards: 70

    Contact: "501"<sip:501@192.168.1.200:5060;transport=udp>

    To: "100"<sip:100@office.temp-art.net:5060>

    From: "501"<sip:501@office.temp-art.net:5060>;tag=03665448cf

    Call-ID: 7038277f10ec7b36

    CSeq: 26241 INVITE

    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO

    Supported: timer, 100rel, replaces

    User-Agent: Aastra 53i/2.2.0.166

    Allow-Events: talk, hold, conference

    Content-Length: 0




    17:07:55.465 [CM302001]: Authorization system can not identify source of: SipReq: INVITE 100@office.temp-art.net:5060 tid=c097d65f23a380942 cseq=INVITE contact=501@192.168.1.200:5060 / 26241 from(wire)

    17:07:45.753 [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:100@192.168.1.215:2051;line=gitvy0v4/100]

    17:07:16.388 [CM504001]: Ext.501: new contact is registered. Contact(s): [sip:501@192.168.1.200:5060;transport=udp/501]

    17:06:58.041 [CM504001]: Ext.400: new contact is registered. Contact(s): [sip:400@192.168.1.215:2051;line=h09qiamt/400]

    17:05:07.052 [CM504001]: Ext.102: new contact is registered. Contact(s): [sip:102@192.168.1.213:2051;line=anmzqt2r/102]

    17:05:01.909 [CM504008]: Fax Service: registered as sip:888@192.168.1.99:5060 with contact sip:888@192.168.1.99:5100;user=phone

    17:03:12.632 [CM504001]: Ext.103: new contact is registered. Contact(s): [sip:103@192.168.1.214:2051;line=ow60ld50/103]

    17:02:45.532 [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:100@192.168.1.215:2051;line=gitvy0v4/100]

    17:01:57.771 [CM504001]: Ext.400: new contact is registered. Contact(s): [sip:400@192.168.1.215:2051;line=h09qiamt/400]

    17:00:01.759 [CM504008]: Fax Service: registered as sip:888@192.168.1.99:5060 with contact sip:888@192.168.1.99:5100;user=phone

    16:59:48.372 [CM503008]: Call(1466): Call is terminated

    16:59:48.370 [CM503008]: Call(1466): Call is terminated

    16:59:35.419 [CM503007]: Call(1466): Device joined: sip:100@192.168.1.215:2051;line=gitvy0v4

    16:59:35.417 [CM503007]: Call(1466): Device joined: sip:anonymous@192.168.1.180:5060

    16:59:33.749 [CM505001]: Ext.103: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:59:33.749 [CM503002]: Call(1466): Alerting sip:103@192.168.1.214:2051;line=ow60ld50

    16:59:33.478 [CM505001]: Ext.100: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:59:33.478 [CM503002]: Call(1466): Alerting sip:100@192.168.1.215:2051;line=gitvy0v4

    16:59:33.183 [CM505001]: Ext.102: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom320/6.5.16] Transport: [sip:192.168.1.99:5060]

    16:59:33.183 [CM503002]: Call(1466): Alerting sip:102@192.168.1.213:2051;line=anmzqt2r

    16:59:33.089 [CM505001]: Ext.101: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:59:33.089 [CM503002]: Call(1466): Alerting sip:101@192.168.1.216:2051;line=z68xfxs0

    16:59:32.896 [CM503024]: Call(1466): Calling RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:102@192.168.1.213:2051;line=anmzqt2r]

    16:59:32.878 [CM503024]: Call(1466): Calling RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:101@192.168.1.216:2051;line=z68xfxs0]

    16:59:32.867 [CM503024]: Call(1466): Calling RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:103@192.168.1.214:2051;line=ow60ld50]

    16:59:32.858 [CM503024]: Call(1466): Calling RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:100@192.168.1.215:2051;line=gitvy0v4]

    16:59:32.849 [CM503004]: Call(1466): Route 1: RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:100@192.168.1.215:2051;line=gitvy0v4, Dev:sip:103@192.168.1.214:2051;line=ow60ld50, Dev:sip:101@192.168.1.216:2051;line=z68xfxs0, Dev:sip:102@192.168.1.213:2051;line=anmzqt2r]

    16:59:32.845 [CM503010]: Making route(s) to <sip:852@192.168.1.99:5060>

    16:59:32.844 [CM505002]: Gateway:[Patton 4114] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4114 JO EUI 00A0BA043086 R5.1 2008-11-12 H323 SIP FXS FXO M5T SIP Stack/4.0.23.23] Transport: [sip:192.168.1.99:5060]

    16:59:32.841 [CM503001]: Call(1466): Incoming call from anonymous@(Ln.10000@Patton 4114) to <sip:852@192.168.1.99:5060>

    16:59:32.835 [CM503012]: Inbound office hours rule (unnamed) for 10000 forwards to DN:852

    16:58:28.369 [CM503008]: Call(1465): Call is terminated

    16:58:28.367 [CM503008]: Call(1465): Call is terminated

    16:58:17.616 [CM503007]: Call(1465): Device joined: sip:100@192.168.1.215:2051;line=gitvy0v4

    16:58:17.614 [CM503007]: Call(1465): Device joined: sip:anonymous@192.168.1.180:5060

    16:58:15.740 [CM505001]: Ext.103: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:58:15.740 [CM503002]: Call(1465): Alerting sip:103@192.168.1.214:2051;line=ow60ld50

    16:58:15.468 [CM505001]: Ext.100: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:58:15.467 [CM503002]: Call(1465): Alerting sip:100@192.168.1.215:2051;line=gitvy0v4

    16:58:15.128 [CM505001]: Ext.102: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom320/6.5.16] Transport: [sip:192.168.1.99:5060]

    16:58:15.128 [CM503002]: Call(1465): Alerting sip:102@192.168.1.213:2051;line=anmzqt2r

    16:58:15.077 [CM505001]: Ext.101: Device info: Device Identified: [Man: Snom;Mod: 3xx series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [snom360/6.5.13] Transport: [sip:192.168.1.99:5060]

    16:58:15.077 [CM503002]: Call(1465): Alerting sip:101@192.168.1.216:2051;line=z68xfxs0

    16:58:14.905 [CM503024]: Call(1465): Calling RingAll852:100Ext.100103Ext.103101Ext.101102Ext.102@[Dev:sip:102@192.168.1.213:2051;line=anmzqt2r]
     
  4. leejor

    leejor Well-Known Member

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    Ok, a couple of incoming calls are going to the ring group, with 4 sets. Ext 100 answers and then hangs up at 16:58:28 and 16:59:48 (calls terminated) at which point you (3CX) still sees the extension in use? 3CX seems to indicate that knows that the call is over. It may be nothing but I'm wondering why you are using port 2051 on the sets rather than the more "standard" 5060, 5061, etc?
     
  5. tpinnovations

    tpinnovations Member

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    Sounds like an RTP issues, Have you made sure all the the proper ports are opened/forwarded on your firewall?
     
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  6. RobertKroll

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    I actually didn't change the ports (intentionally at least). Do you think that could be an issue?
     
  7. tpinnovations

    tpinnovations Member

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    It would be best if you reset everything to the default port of 5060.
     
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  8. LeonidasG

    LeonidasG Support Team
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    There may be an issue with using another Port other than 5060... It's a possibility.
    So just try and use the Standard 5060 Port and see if your problem is solved.

    While you're doing this i'll try and reproduce your issue on my machine and see what happens.
     
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  9. RobertKroll

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    I can't see where I can change the port. I logged into the phone (SNOM 360), but I can't find the location. Please excuse my ignorance, I am hanging on by a thread here (I don't do this for a living, and it's a good thing)! I found a few other settings, please tell me what you think about these?

    Auto Answer: on off
    Long SIP-Contact (RFC3840): on OFF
    Support broken Registrar: ON off
    Shared Line: on OFF
    DTMF via SIP INFO: on OFF
    Send display name on INVITE: on OFF

    I was wondering if one of these may have bearing on the problem. Oddly though, we have had the system for about a year, and we have only experienced this particular error for the past 3-4 months.
     
  10. leejor

    leejor Well-Known Member

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    If a system had been working, up to a certain point, and then started having problems, then something had to have changed. It's usually something that people can't possibly conceive as making a difference, but it can, and does. I would try to think back and figure out what was changed/added on your network just before you began having the troubles.
     
  11. RobertKroll

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    The only change we made was upgrading from the Release Candidate of V.7 to the full release. That is when the issues basically started happening. I cannot locate a spot to change the port #. I tried resetting the system and the Patton gateway. The problem seems to be intermittent. It may happen once today, and 30 times tomorrow. I cannot seem to find a pattern. I dropped a message to Matt Landis, but I have not yet heard back from him. Some of the other things we tried didn't work for us either, and I think it may be time for me to bring in a pro.

    On another note, I keep getting this "INVITE" error when trying to connect with my remote phone. It registers, but calls fail, I have read just about every post in the forums, but I cannot seem to repair this. I have opened all of the recommended ports (I believe). As I said, It is probably time to bring a pro in.

    Kindest regards and appreciation for your help.
    Bob
     
  12. Nick Galea

    Nick Galea Site Admin

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    Hi Robert,

    This is due to the fact that your gateway is not detecting the hang up. You need to ensure that the gateway is correctly configured for your country - as the tone sets will differ.

    There also seems to be an issue with your patton configuration. Calls from your gateway are not being identified.

    3CX Support can help you configure the patton for your country providing you have a commercial edition and support package or are a 3CX Partner.
     
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  13. RobertKroll

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    Can his have something to do with calls from extension to extension and extension to voicemail remaining active? Outside calls seem to perform correctly across the board. I also would like to know where to change the port as discussed earlier...I would at least like to get that back on track as you suggested. What is the best way to get support? Should I go through a third party (Landis computer) or do I go through 3cx directly?

    Regards,
    Bob
     
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