Forward extension to direct sip

Discussion in '3CX Phone System - General' started by mjmcfarl, Feb 23, 2012.

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  1. mjmcfarl

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    Using softphone am able to call to a direct sip accout (username@iptel.org) when registered to 3cx. Would like to be able to set forwarding rules on an extension so that if extension is not answered or not connected can forward to the username@iptel.org account. Is there a format of the direct sip address which can be used to forward an extension to?

    Thanks.

    Matt
     
  2. leejor

    leejor Well-Known Member

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    Create a new trunk group for iptel.org, set it to not register, in fact many of the settings (password/user) are not critical. You just want to create a conduit to that URL. In your outbound rules, create a rule that sends the required digits to that trunk group. Depending on your current dialplan and possible conflicts, you may want to "convert" a longer number, by stripping all digits and then prepending only the required digits, before sending out.
     
  3. mjmcfarl

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    Setup the trunk and outbound rules, but still not getting processed correctly. Included snippet from the logs showing a successful call through 3cx using direct sip dialing and the failed call which is placed to an internal extension with the phone unregistered and set to forward.

    Thanks for your help.

    Failed call from 650 to unregistered 651 setup to forward to <sip:my_username@iptel.org>
    14:22:21.003 [CM503020]: Normal call termination. Reason: Terminated
    14:22:21.003 [CM503016]: Call(123): Attempt to reach <sip:651@192.168.3.14> failed. Reason: Not Acceptable HereReason Unknown
    14:22:21.003 [CM503003]: Call(123): Call to sip:%3Csip%3Amy_username%40iptel.org%3E@iptel.org:5060 has failed; Cause: 407 Proxy Authentication Required; warning: Noisy feedback tells: pid=29557 req_src_ip=xxx.xxx.xxx.xxx req_src_port=62077 in_uri=sip:%3Csip%3Amy_username%40iptel.org%3E@iptel.org:5060 out_uri=sip:%3Csip%3Amy_username%40iptel.org%3E@iptel.org:5060 via_cnt==1; from IP:xxx.xxx.xxx.xxx:5060
    14:22:17.894 [CM503025]: Call(123): Calling VoIPline:<sip:my_username@iptel.org>@(Ln.10000@iptel-trunk)@[Dev:sip:my_username@iptel.org:5060]
    14:22:16.972 [CM503005]: Call(123): Forwarding: VoIPline:<sip:my_username@iptel.org>@(Ln.10000@iptel-trunk)@[Dev:sip:my_username@iptel.org:5060]
    14:22:16.972 [CM503016]: Call(123): Attempt to reach <sip:651@192.168.3.14> failed. Reason: Not Registered
    14:22:16.972 [CM503017]: Call(123): Target is not registered: Ext.651
    14:22:16.972 [CM503010]: Making route(s) to <sip:651@192.168.3.14>
    14:22:16.972 [CM505001]: Ext.650: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite 4 release 4.1 stamp 63214] PBX contact: [sip:650@192.168.3.14:5060;transport=TCP]
    14:22:16.972 [CM503001]: Call(123): Incoming call from Ext.650 to <sip:651@192.168.3.14>


    Successful call from 650 to my_username@iptel.org.
    14:28:10.695 [CM503007]: Call(125): Device joined: sip:my_username@iptel.org
    14:28:10.695 [CM503007]: Call(125): Device joined: sip:650@192.168.0.100:54622;transport=TCP
    14:28:07.211 [CM505001]: Ext.my_username: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:my_username@xxx.xxx.xxx.xxx:62140]
    14:28:07.211 [CM503002]: Call(125): Alerting sip:my_username@iptel.org
    14:28:02.211 [CM503025]: Call(125): Calling Ext:Sip.DirectSIP@[Dev:sip:my_username@iptel.org]
    14:28:01.273 [CM503004]: Call(125): Route 1: Ext:Sip.DirectSIP@[Dev:sip:my_username@iptel.org]
    14:28:01.273 [CM503010]: Making route(s) to <sip:my_username@iptel.org>
    14:28:01.273 [CM505001]: Ext.650: Device info: Device Identified: [Man: Counterpath;Mod: X-Lite;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [X-Lite 4 release 4.1 stamp 63214] PBX contact: [sip:650@192.168.3.14:5060;transport=TCP]
    14:28:01.273 [CM503001]: Call(125): Incoming call from Ext.650 to <sip:my_username@iptel.org>
     
  4. leejor

    leejor Well-Known Member

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    It appears that the call is being sent out.

    But, the far end does not like what it is getting..

    I'm not familiar with iptel parameters so I can't help you on why it is not completing, perhaps someone else on the forum has used them before and can comment. You are reaching them, so your outbound rules are working correctly.

    I'm wondering if 3CX is sending along caller ID/user information that is not acceptable, because of the call forwarding?
     
  5. Anonymous

    Anonymous Guest

    We too have tried serval tricks to accomplish this via various methods including call API and header modifications, but to no avail.

    Turns out that this is not possible with 3CX, at least at the moment.
     
  6. leejor

    leejor Well-Known Member

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    I suppose, if you wish to pursue this, you could "Wireshark", a direct, and a forwarded call, to compare the difference (exactly what is being sent in each case). It might give some insight into where to go next.
     
  7. mjmcfarl

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    Thanks for the suggestion. There were differences in the sip-uri, to and from fields. I was able to identify them using wireshark. Then was able to use the Outbound Parameters setting on the trunk config to adjust them properly.
     
  8. Anonymous

    Anonymous Guest

    Were you then able to make a direct sip call?
     
  9. fujiconsulting

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    I am not having any luck, did this ever get solved??

    Here is the output i get when placing a call that i have tried to forward to a direct sip address in the form of number (*********)@sip1.didx.net. I have noticed that for some reason the PBX turns the @ to a %?? But here is the output from the server activity log:

    23:10:02.922 [CM503007]: Call(3): Device joined: sip:10014@127.0.0.1:5066
    23:10:02.744 [CM503025]: Call(3): Calling Unknown:Ext.EndCall@[Dev:sip:EndCall@127.0.0.1:40600;rinstance=93c2f8ebd0dc12ce]
    23:10:02.724 [CM503016]: Call(3): Attempt to reach <sip:**********%40sip1.didx.net@192.168.200.16:5060> failed. Reason: Not Found
    23:10:02.714 [CM503014]: Call(3): No known route to target: <sip:**********%40sip1.didx.net@192.168.200.16:5060>
    23:10:02.708 [CM503001]: Call(3): Incoming call from 9076323854@(Ln.10014@ACS T1) to <sip:**********%40sip1.didx.net@192.168.200.16:5060>
    23:10:02.691 [CM503012]: Inbound any hours rule (unnamed) for 10014 forwards to DN:**********@sip1.didx.net

    Thanks!
     
  10. leejor

    leejor Well-Known Member

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    Are you manually adding the @ (in the sent digits rules or the trunk datafill) , or just letting 3CX insert it itself before sending out the number?
     
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