Multiple external numbers as part of RingAll Ring group

Discussion in '3CX Phone System - General' started by scruffybob, Nov 24, 2009.

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  1. scruffybob

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    I think this is very similar to a recent thread, but I didn't want to take over anyone elses thread, so I'll explain my problem.

    Scenario
    When our office administrator is busy, we want to automatically roll over calls to our cell phones over outgoing analog lines. We'd like to ring two cell phones simultaneously, and let the first one that can take the call pick it up. If neither cell phone is answered, we'd like the call to go to our internal email.

    Setup
    Hardware: Grandstream GXW-4104 handling the 4 analog lines (only 3 analog lines are connected for this test although all four have been provisioned on the 4104 - 1 is used for the incoming call, 2 should be needed for the two outgoing calls). We haven't had any issues receiving or sending any other calls and are pretty much configured as indicated in the 3cx configuration guides, so I'm not inclined to think this is a gateway setup issue. All the gateways are flashed to the latest software releases.

    Software: 3cx (free version, we're still testing to make sure this will do what we need it to do) release 8.0.9532.468. We can make outgoing calls to external numbers from the office, we can ForwardAll from an extension to an external number with no issue.

    Configuration
    The ring group:
    Virtual number: 801
    Ring Strategy: Ring All We want them to all ring simultaneously
    Ring Time: 10 (We'd like to pull the call back if it's not answered quickly)
    Extensions 100 and 101 are members

    The extensions:
    Extension numbers: 100 & 101
    Forward ALL calls / Out of Office is checked
    An external number of Skype ID is filled in with correct external phone numbers for each extension

    Outbound Rules:
    There is only one outbound rule:
    Route to the GXW-4104
    Calls to numbers with length 11 (e.g. 1-630-555-1212)
    Strip 0 digits

    In the 3cX PSTN setup for the 4104:
    Starts with line 10000 (so the analog lines are numbered 10000-10003)
    PBX delivers audio is checked (we use this because we were having problems with getting voice on hairpinned calls on the 4104)

    Global Options:
    Allow forwarding to external numbers in ring groups is set

    Let me know if you need anymore configuration information. (I've edited the phone numbers for security purposes):

    Here is the relevant part of the logs that show the call flow:
    Code:
    21:59:45.925  [CM503007]: Call(121): Device joined: sip:10003@192.168.2.162:5066;transport=udp
    21:59:45.909  [MS210002] C:121.1:Offer provided. Connection(transcoding mode): 192.168.2.160:7258(7259)
    21:59:45.894  [MS210001] C:121.4:Answer received. RTP connection[unsecure]: 192.168.2.162:5016(5017)
    21:59:45.894  Remote SDP is set for legC:121.4
    21:59:45.862  [CM505002]: Gateway:[Analog Trunks - GXW-4104] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 1.1, Ch:11) 1.2.1.5] PBX contact: [sip:10003@192.168.2.160:5060;transport=udp]
    21:59:45.862  [CM503002]: Call(121): Alerting sip:10002@192.168.2.162:5064;transport=udp
    21:59:42.566  [CM503003]: Call(121): Call to sip:16308860000@192.168.2.162:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.2.162:5060
    21:59:42.550  [CM503003]: Call(121): Call to sip:16308860000@192.168.2.162:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.2.162:5060
    21:59:42.534  [CM503003]: Call(121): Call to sip:16308860000@192.168.2.162:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.2.162:5060
    21:59:42.519  [CM503003]: Call(121): Call to sip:16304530000@192.168.2.162:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.2.162:5060
    21:59:42.456  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10003@192.168.2.162:5066;transport=udp]
    21:59:42.456  [MS210002] C:121.8:Offer provided. Connection(transcoding mode): 192.168.2.160:7272(7273)
    21:59:42.441  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10002@192.168.2.162:5064;transport=udp]
    21:59:42.425  [MS210002] C:121.7:Offer provided. Connection(transcoding mode): 192.168.2.160:7270(7271)
    21:59:42.409  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10001@192.168.2.162:5062;transport=udp]
    21:59:42.409  [MS210002] C:121.6:Offer provided. Connection(transcoding mode): 192.168.2.160:7268(7269)
    21:59:42.394  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10003@192.168.2.162:5066;transport=udp]
    21:59:42.378  [MS210002] C:121.5:Offer provided. Connection(transcoding mode): 192.168.2.160:7266(7267)
    21:59:42.362  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10002@192.168.2.162:5064;transport=udp]
    21:59:42.347  [MS210002] C:121.4:Offer provided. Connection(transcoding mode): 192.168.2.160:7264(7265)
    21:59:42.331  [CM503025]: Call(121): Calling RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10001@192.168.2.162:5062;transport=udp]
    21:59:42.316  [MS210002] C:121.3:Offer provided. Connection(transcoding mode): 192.168.2.160:7262(7263)
    21:59:42.284  [CM503004]: Call(121): Route 1: RingAll801:10016304530000@(Ln.10001@Analog Trunks - GXW-4104)10116308860000@(Ln.10001@Analog Trunks - GXW-4104)@[Dev:sip:10001@192.168.2.162:5062;transport=udp,Dev:sip:10002@192.168.2.162:5064;transport=udp,Dev:sip:10003@192.168.2.162:5066;transport=udp,Dev:sip:10001@192.168.2.162:5062;transport=udp,Dev:sip:10002@192.168.2.162:5064;transport=udp,Dev:sip:10003@192.168.2.162:5066;transport=udp]
    21:59:42.253  [CM503010]: Making route(s) to <sip:801@127.0.0.1:5060>
    21:59:42.253  Refer: from=<sip:800@127.0.0.1:5060>;tag=a96d6132; to="M, RON E"<sip:6303570000@127.0.0.1:5060>;tag=d23e5003; RefTo=<sip:801@127.0.0.1:5060>
    
    So, in the end we have two issues:
    1. Only one external number ever rings (in the case above, it was the 1-630-453-0000 number. The second number never rings.
    2. The call is never pulled back. Regardless of how low the "Ring Time" is set in the definition of group 801, the call continues to ring to the one number.

    So, am I missing something in the configuration, or am I trying to do something that can't be done with 3cx? It would seem like this would be a pretty basic thing to want to do (forward a call to several external numbers if an incoming call isn't handled by the onsite administrator), so I'm surprised at how hard it's been to configure this properly.

    Extra credit: :lol:
    I alluded to another problem we've had with the Grandstream 4104. When we have an incoming call come into the 4104 and we forward it to an external number with the 3cx, the 3cx will appropriately hand off both ends to the 4104 and take itself out of the loop. Unfortunately, when it does that, the call connects just fine, but neither end can hear the other. I fixed the problem by checking "PBX delivers Audio" box in the 3cx PSTN configuration, so all the audio goes through the 3cx, but this seems to be inefficient. This seems to be a known problem, but was just wondering if there was any workaround for this yet.

    Best regards,

    scruffy-bob
     
  2. leejor

    leejor Well-Known Member

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    Does a simple call to the ring group containing the extensions forwarded to the mobiles, ring both mobiles? If not, then there is no point continuing, there could be an issue with two outside calls at the same time perhaps. Break whole string of events down, try a single call to the extension forwarded to the mobile, in other words what happens if you take one of the extensions out of the ring group?
     
  3. scruffybob

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    As I indicated above, we can make simple calls to external numbers (internal to external), and the ForwardAll works by forwarding a call to an internal extension to an external number. These are no problem at all and work as expected.

    When we take one of the two people out of the ring group (leaving a ring group of 1), it has no problem dialing the single outside number.

    It's only dialing two external numbers simultaneously that seems to fail.

    The GWX-4001 can dial multiple outgoing numbers simultaneously (two internal calls to two external calls).

    I just don't understand why it would be any different dialing two external numbers versus two internal numbers. I'm still thinking I haven't set something up correctly or it would all work.

    [By the way, I worked around the problem temporarily by using my Google Voice to do the same thing so I can finish my testing. Essentially, my ForwardAll external call goes to my Google Voice account that then does a simultaneous ring to the separate numbers. I don't want to even think about the number of ways that could stop working at any moment, but seems to work fine right now.]

    Best regards,
    scruffy-bob
     
  4. leejor

    leejor Well-Known Member

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    Since it seems to work with just the one extension in the ring group, and this is just a guess, perhaps the way the call is passing through 3CX it sees the second outgoing call forward as more than the 4 call limit in the free edition. Perhaps someone that has a licensed version with a higher limit can see if this is the case, or, it is 3CX that is having a problem doing the simultaneous outside call forward.
     
  5. bobo8

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    I have 2 external numbers (mobile phones) in the Ring group, they ring properly. I have free 3CX, ver. 8.0.9908.
     
  6. wzaatar

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    In you GXW-410x, how did you configure your Port Scheduling Schema (Voip->PSTN)?
     
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  7. scruffybob

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    Left it at the default:
    Code:
    Port Scheduling Schema (Voip->PSTN)    rr:1-4;
      (Syntax: rr: port_group; [...]) 
      (Default: rr:1-4; round-robin of all ports ) 
    
    scruffy-bob
     
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