[asterisk-users] SEMI-OFFTOPIC openvox

ricky gutierrez xserverlinux at gmail.com
Mon Jan 19 14:37:34 CST 2015


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

Contact: <sip:984783842 at 50.X.X.X:5060>
Content-Length: 0


<------------>
    -- Executing [984783842 at 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 at 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" <sip:101 at 50.X.X.X>;tag=as3708c762
To: <sip:84783842%40 at 190.53.38.203>
Contact: <sip:101 at 50.X.X.X:5060>
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 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"
<sip:101 at 50.X.X.X>;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" <sip:101 at 50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a at 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: <sip:984783842 at 50.X.X.X:5060>
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" <sip:101 at 50.X.X.X>;tag=as3708c762
To: <sip:84783842%40 at 190.53.38.203>;tag=as4bb74f30
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 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 at 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" <sip:101 at 50.X.X.X>;tag=as3708c762
To: <sip:84783842%40 at 190.53.38.203>;tag=as4bb74f30
Contact: <sip:101 at 50.X.X.X:5060>
Call-ID: 0c9236b922c5a99f6a1a797c7c3f9eb7 at 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" <sip:101 at 50.X.X.X>;tag=as3708c762'
Scheduling destruction of SIP dialog
'0c9236b922c5a99f6a1a797c7c3f9eb7 at 50.X.X.X:5060' in 32000 ms (Method:
INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [984783842 at 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" <sip:101 at 50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a at 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 at 50.X.X.X SIP/2.0
Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36
From: "101" <sip:101 at 50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842 at 50.X.X.X>;tag=as30070ac7
Call-ID: 7f55e32e-e4c6e11a at 172.16.8.179
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101 at 190.X.X.1:41316>
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" <sip:101 at 50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842 at 50.X.X.X>;tag=as77fb37e2
Call-ID: 7f55e32e-e4c6e11a at 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 <xserverlinux at gmail.com>:
> 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



-- 
rickygm

http://gnuforever.homelinux.com



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