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nebbin

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Posts posted by nebbin

  3. SBC appliance

    in General Setup

    Posted

    We often get stuck with issues concerning registrations to our Pbx server with customers with difficult environment. 3cx has a SBC appliance/software they put on site to handle these situations and to ensure secure connection back to hosted PBX. Do we have anything like this for Vodia.

    We lost a client due to issues with their firewall/router specialized appliance used to daisy chain phone and computers. Wouldn't  allow us access to this appliance or put phones IP's in DMZ. Registrations were dropping due to clients appliance.

    NE

  5. Dialing an Extension in another domain in the PBX

    in Call Treatment

    Posted

    Seemed to figured it out. The auto attendant number to select extension was being sent out directly on trunk to FreePBX, but putting pound sign after the number on the auto attendant, this behavior stopped, i.e. isn't set out as DMTF input over trunk. Trunk was expecting a PIN number...so if waited for the initial entry period to expire (i.e. incomplete input), then second as for input - hence initial delay for external calls.

    NE

  7. Dialing an Extension in another domain in the PBX

    in Call Treatment

    Posted

    Here's the log from the calls the delay (up to 40 seconds) media. It is because if trying to find where to place call:

    In this case I'm call from a cell phone (4415362338)  to the conference room number via 1 441 400 7000 and then extension 4000:

    [8] 11:13:26.917 TRUN: Trying to match number 14412362095 with ERE (.*)
    [8] 11:13:26.917 TRUN: Send call to extension ERE returned 14412362095
    [8] 11:13:26.919 TRUN: Trying to match number 14412362095 with ERE (.*)
    [8] 11:13:26.919 TRUN: Send call to extension ERE returned 14412362095
    [8] 11:13:26.919 TRUN: Trying to match number 14412362095 with ERE (.*)
    [8] 11:13:26.919 TRUN: Send call to extension ERE returned 14412362095
    [8] 11:13:26.919 TRUN: Trying to match number 14412362095 with ERE (.*)
    [8] 11:13:26.919 TRUN: Send call to extension ERE returned 14412362095
    [8] 11:13:26.920 TRUN: Trying to match number 14412362095 with ERE (.*)
    [8] 11:13:26.920 TRUN: Send call to extension ERE returned 14412362095
    [9] 11:13:47.367 TRUN: Generating hf header using <sip:{ext-ani}@{domain}>
    [9] 11:13:47.367 TRUN: Generating ht header using {to}
    [9] 11:13:47.367 TRUN: Generating hpai header using <sip:{ext-ani}@{domain}>
    [9] 11:13:47.367 TRUN: Generating hrpi header using {from}
    [9] 11:13:48.048 TRUN: Generating hf header using <sip:{ext-ani}@{domain}>
    [9] 11:13:48.048 TRUN: Generating ht header using {to}
    [9] 11:13:48.048 TRUN: Generating hpai header using <sip:{ext-ani}@{domain}>
    [9] 11:13:48.048 TRUN: Generating hrpi header using {from}
    [8] 11:13:56.107 TRUN: Trying to match number 14414055308 with ERE (.*)
    [8] 11:13:56.107 TRUN: Send call to extension ERE returned 14414055308
    [8] 11:13:56.108 TRUN: Trying to match number 14414055308 with ERE (.*)
    [8] 11:13:56.108 TRUN: Send call to extension ERE returned 14414055308
    [8] 11:13:56.109 TRUN: Trying to match number 14414055308 with ERE (.*)
    [8] 11:13:56.109 TRUN: Send call to extension ERE returned 14414055308
    [8] 11:13:56.109 TRUN: Trying to match number 14414055308 with ERE (.*)
    [8] 11:13:56.109 TRUN: Send call to extension ERE returned 14414055308
    [8] 11:13:56.109 TRUN: Trying to match number 14414055308 with ERE (.*)
    [8] 11:13:56.109 TRUN: Send call to extension ERE returned 14414055308
    [9] 11:14:01.342 TRUN: Generating hf header using <sip:{ext-ani}@{domain}>
    [9] 11:14:01.343 TRUN: Generating ht header using {to}
    [9] 11:14:01.343 TRUN: Generating hpai header using <sip:{ext-ani}@{domain}>
    [9] 11:14:01.343 TRUN: Generating hrpi header using {from}
    [9] 11:14:02.198 TRUN: Generating hf header using <sip:{ext-ani}@{domain}>
    [9] 11:14:02.198 TRUN: Generating ht header using {to}
    [9] 11:14:02.198 TRUN: Generating hpai header using <sip:{ext-ani}@{domain}>
    [9] 11:14:02.198 TRUN: Generating hrpi header using {from}
    [5] 11:14:10.416 PACK: SIP Rx 76.8.40.106:5060: 
    BYE sip:14414007000@172.24.16.181:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 76.8.40.106:5060;branch=z9hG4bK-524287-1---2aec38613f440f6ee40ff213f9227b6e;rport
Via: SIP/2.0/UDP 76.8.40.108:5070;rport=5070;branch=z9hG4bK-e2lwqbk6uo5yvqna
Max-Forwards: 69
Contact: sip:76.8.40.108:5070
To: <sip:14414007000@172.24.16.181>;tag=9a1a289d78
From: <sip:14415362338@76.8.40.106>;tag=CQFRGNYRWBOVWOCPGJMA____.o
Call-ID: 0gQAAC8WAAACBAAALxYAABrTWpr/c1LmFESM7HBU77DJ/8mO4EoN/uALHlILOlZV@69.17.214.94
CSeq: 854 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
User-Agent: PortaSIP
cisco-GUID: 3753021899-3943194721-784679244-1290984961
h323-conf-id: 3753021899-3943194721-784679244-1290984961
Content-Length: 0
    [5] 11:14:10.417 PACK: SIP Tx 76.8.40.106:5060: 
    SIP/2.0 200 Ok
Via: SIP/2.0/UDP 76.8.40.106:5060;branch=z9hG4bK-524287-1---2aec38613f440f6ee40ff213f9227b6e;rport=5060
Via: SIP/2.0/UDP 76.8.40.108:5070;rport=5070;branch=z9hG4bK-e2lwqbk6uo5yvqna
From: <sip:14415362338@76.8.40.106>;tag=CQFRGNYRWBOVWOCPGJMA____.o
To: <sip:14414007000@172.24.16.181>;tag=9a1a289d78
Call-ID: 0gQAAC8WAAACBAAALxYAABrTWpr/c1LmFESM7HBU77DJ/8mO4EoN/uALHlILOlZV@69.17.214.94
CSeq: 854 BYE
Contact: <sip:14414007000@204.232.169.181:5060;transport=udp>
User-Agent: Vodia-PBX/65.0.6
Content-Length: 0
    [5] 11:14:10.426 PACK: SIP Tx 155.138.208.200:5060: 
    BYE sip:8990@155.138.208.200:5060 SIP/2.0
Via: SIP/2.0/UDP 204.232.169.181:5060;branch=z9hG4bK-edfcf87d38e9c0536007dbc7f4d126af;rport
From: <sip:14414007000@pbx.gombay.bm>;tag=939817284
To: "Gombay Main Auto attendant" <sip:8990@pbx.gombay.bm;user=phone>;tag=as6dfb3053
Call-ID: 8f7da753@pbx
CSeq: 20696 BYE
Max-Forwards: 70
Contact: <sip:4000@204.232.169.181:5060;transport=udp>
P-Asserted-Identity: "YAK BM" <sip:4000@pbx.gombay.bm>
P-Preferred-Identity: "YAK BM" <sip:4000@pbx.gombay.bm>
Remote-Party-ID: "YAK BM" <sip:14414007000@>
Content-Length: 0
    [5] 11:14:10.439 PACK: SIP Rx 155.138.208.200:5060: 
    SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.16.181:5060;branch=z9hG4bK-edfcf87d38e9c0536007dbc7f4d126af;received=172.24.16.181;rport=5060
From: <sip:14414007000@pbx.gombay.bm>;tag=939817284
To: "Gombay Main Auto attendant" <sip:8990@pbx.gombay.bm;user=phone>;tag=as6dfb3053
Call-ID: 8f7da753@pbx
CSeq: 20696 BYE
Server: FPBX-15.0.16.53(16.9.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
    [7] 11:14:10.439 SIP: Port 949: Clear last request
    [5] 11:14:10.439 SIP: BYE Response: Terminate 8f7da753@pbx

     

  8. Dialing an Extension in another domain in the PBX

    in Call Treatment

    Posted

    Note: in this case, the conference is tied first in FreePBX (where the IVR integration takes place because we don't know how to do integration to conference App in Vodia). So we're forwarding the calls to FreePBX? Should we then not be using and extension DID in Vodia, but a trunk with a DID reference. Currently, we set up the DID in the conference domain, to be forward to a number which is identified as Trunk A in a dial plan? Are suggesting we can skip this and send directly to trunk identified as conference DID?

     

    Also, we seem to have gotten Vodia to forward call into this conference (via FreePBX). Calling from a number set up as DID on our system seems to works. But calling from an external number (mobile); it would appear there is no sound for 40 seconds before FreePBX IVR can be heard. What would cause this?

  10. Dialing an Extension in another domain in the PBX

    in Call Treatment

    Posted

    We're offering video conferencing to customers importantly with local dial in number capability. Is it possible to send a conference dial in number to the extension our of conference facility PBX domain from an extension in a customer domain or from their auto attendant (DID). We could place a DID on the our extension in the conference facility domain connected to the video conference facility, but I think we want the conference facility extension to dial out to avoid carrier concurrent call restrictions on DID.

    Any ideas on how we could approach this?

    Thanks,

    NE

  13. Call Center - Call Automation

    in Call Treatment

    Posted

    We currently using Vodia PBX with our client's middle software that is voice automation + a bit of AI in the transportation business. For a current call, what API can we use to  call an external number to dial and external number and connect this external to a  current call to of an agent?

    The business scenario in this case is to connect a incoming call to the number (cell phone) of a transport driver, without any human intervention. The middleware developed will queue it's own database to determine the appropriate number to dial. However, with this information what Vodia commands (API)/code do we need use to achieve this and are there any examples of such code.

    Thanks,

    NE

  17. BLF - NW BUTTONS V60.00

    in General Setup

    Posted

    Seems like there is NO combined BLF for monitor, pick up and speed dial. Client has SNOM 320's and only has 10 free buttons?

    In Future how can we get around this... We don't have any documentation on new buttons set up, so does this mean 30 buttons will be needed,, when previously on 10 were required to do three function for one extension (button)?

  22. Cell No Longer Showing Extension, but CID

    in Extension Setup

    Posted

    We just moved to 60 and clients says calls redirected to cell phone show CID (which one would expect). They said previously it showed the extension call the cell phone and that's what they want to show?

    Is this now possible and, if so, what setting do we need to change to achieve this for a specific customer (in their own domain).

     

    Thanks,

     

    NE

  25. Voice to Text Service/Integration

    in New Features/Versions

    Posted

    Voicemail-to-text will help us distinguish ourselves from other providers (something busy client would prefer).

    Does Vodia have any plans to assist users to achieve this aim?

    Specifically, something in product or a seamless integration to third-party solution. For example:

    Asterisk, Metswitch, Freeswitch,SipWise, etc all integrate with Mutare's giSTT email gateway (which has wav file to text).

     

    NE

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