SEMI-OFFTOPIC Openvox

Home » Asterisk Users » SEMI-OFFTOPIC Openvox
Asterisk Users 2 Comments

Hi list, I write on the list looking for help, buy a openvox gw gsm for four channels and I’m a little disappointed with the support openvox, for some reason , The call doesn´t get trough

support tells me it was my asterisk server, but does not really work me and my internal calls are working perfectly, I tested with another sangoma FXO gateway and works perfectly.

the problem is that support openvox is Chinese and the difference in time zone is high.

my trunk is connected

5001/5001 X.X.X.X D Yes
Yes 5060

Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline]

I follow this guide , but not work

http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf


rickygm

http://gnuforever.homelinux.com

2 thoughts on - SEMI-OFFTOPIC Openvox

  • Hi, when I make an outgoing call sends me a busy here, and no one is making call

    Contact:
    Content-Length: 0

    < ------------>
    — Executing [984783842@to_pstn:1] Dial(“SIP/101-0000004e”,
    “SIP/5001/84783842@,40,rRT”) in new stack
    == Using SIP VIDEO TOS bits 136
    == Using SIP VIDEO CoS mark 6
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    Audio is at 13780
    Video is at 50.X.X.X:18488
    Adding codec 100003 (ulaw) to SDP
    Adding codec 100004 (alaw) to SDP
    Adding video codec 200004 (h264) to SDP
    Adding video codec 200003 (h263p) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 190.53.38.203:5060:
    INVITE sip:84783842%40@190.53.38.203 SIP/2.0
    Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport Max-Forwards: 70
    From: “Operadora” ;tag=as3708c762
    To:
    Contact:
    Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
    CSeq: 102 INVITE
    User-Agent: inmaconsa-Voice-Sip-ipbx Date: Mon, 19 Jan 2015 20:17:52 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer Remote-Party-ID: “Operadora”
    ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 507

    v=0
    o=root 541548714 541548714 IN IP4 50.X.X.X
    s=inamaconsa-Voice-Sip-pbx c=IN IP4 50.X.X.X
    b=CT:384
    t=0 0
    m=audio 13780 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv m=video 18488 RTP/AVP 99 98
    a=rtpmap:99 H264/90000
    a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
    a=rtpmap:98 H263-1998/90000
    a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
    a=sendrecv


    — Called SIP/5001/84783842@

    < --- Transmitting (NAT) to 190.X.X.1:41316 --->
    SIP/2.0 180 Ringing Via: SIP/2.0/UDP
    190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
    From: “101” ;tag=35721c1e3f767ceao4
    To: ;tag=as77fb37e2
    Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
    CSeq: 102 INVITE
    Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer Contact:

    Content-Length: 0

    < ------------>

    < --- SIP read from UDP:190.53.38.203:5060 --->
    SIP/2.0 403 Forbidden Via: SIP/2.0/UDP
    50.X.X.X:5060;branch=z9hG4bK374c2247;received=50.X.X.X;rport=5060
    From: “Operadora” ;tag=as3708c762
    To: ;tag=as4bb74f30
    Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
    CSeq: 102 INVITE
    Server: VoxStack Wireless Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer Content-Length: 0

    < ------------->
    — (10 headers 0 lines) —
    Transmitting (NAT) to 190.53.38.203:5060:
    ACK sip:84783842%40@190.53.38.203 SIP/2.0
    Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport Max-Forwards: 70
    From: “Operadora” ;tag=as3708c762
    To: ;tag=as4bb74f30
    Contact:
    Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060
    CSeq: 102 ACK
    User-Agent: inmaconsa-Voice-Sip-ipbx Content-Length: 0


    [Jan 19 14:17:53] WARNING[11596][C-0000003d]: chan_sip.c:23037
    handle_response_invite: Received response: “Forbidden” from
    ‘”Operadora” ;tag=as3708c762′
    Scheduling destruction of SIP dialog
    ‘0c9236b922c5a99f6a1a797c7c3f9eb7@50.X.X.X:5060’ in 32000 ms (Method:
    INVITE)
    == Everyone is busy/congested at this time (1:0/0/1)
    — Executing [984783842@to_pstn:2] Busy(“SIP/101-0000004e”, “3”)
    in new stack

    < --- Reliably Transmitting (NAT) to 190.X.X.1:41316 --->
    SIP/2.0 486 Busy Here Via: SIP/2.0/UDP
    190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
    From: “101” ;tag=35721c1e3f767ceao4
    To: ;tag=as77fb37e2
    Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
    CSeq: 102 INVITE
    Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21
    Content-Length: 0

    < ------------>
    == Spawn extension (to_pstn, 984783842, 2) exited non-zero on
    ‘SIP/101-0000004e’

    < --- SIP read from UDP:190.X.X.1:41316 --->
    ACK sip:984783842@50.X.X.X SIP/2.0
    Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36
    From: “101” ;tag=35721c1e3f767ceao4
    To: ;tag=as30070ac7
    Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
    CSeq: 101 ACK
    Max-Forwards: 70
    Contact: “101”
    User-Agent: Cisco/SPA508G-7.5.6
    Content-Length: 0

    < ------------->
    — (10 headers 0 lines) —
    Retransmitting #1 (NAT) to 190.X.X.1:41316:
    SIP/2.0 486 Busy Here Via: SIP/2.0/UDP
    190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316
    From: “101” ;tag=35721c1e3f767ceao4
    To: ;tag=as77fb37e2
    Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
    CSeq: 102 INVITE
    Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21
    Content-Length: 0

    2015-01-19 10:24 GMT-06:00 ricky gutierrez :


    rickygm

    http://gnuforever.homelinux.com

  • I’ve had some experience with OpenVox GSM cards and chan_extra. Their support isn’t great; they like if you can give them ssh access to your box, and you will need to ask questions afterwards to find out what they did in there, but they did manage to sort out an obscure problem for me and explained enough for me to work out what had been the matter in the first place.

    As far as I can work out, their GSM gateway appliances seem to be some kind of server motherboard with GSM cards and a pre-installed Linux, Asterisk and chan_extra; but I’ve not had direct experience of them, having built my own boxes using G400P and/or G400E cards in my favourite supplier’s motherboards.

    Oh, and finally, if you’re using any kind of GSM gateway, be careful!
    Otherwise, you will end up incurring the wrath of your telco — “unlimited”
    often does not really mean unlimited, and the only way to find out what the limit actually is is to exceed it.


    AJS

    Note: Originating address only accepts e-mail from list! If replying off-
    list, change address to asterisk1list at earthshod dot co dot uk .