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xserverlinux at gmail.com Guest
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Posted: Mon Jan 19, 2015 11:25 am Post subject: [asterisk-users] SEMI-OFFTOPIC openvox |
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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
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xserverlinux at gmail.com Guest
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Posted: Mon Jan 19, 2015 3:38 pm Post subject: [asterisk-users] SEMI-OFFTOPIC openvox |
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Hi, when I make an outgoing call sends me a busy here, and no one is making call
Contact: <sip:984783842@50.X.X.X:5060>
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" <sip:101@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>
Contact: <sip:101@50.X.X.X:5060>
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"
<sip:101@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@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;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: <sip:984783842@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@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>;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" <sip:101@50.X.X.X>;tag=as3708c762
To: <sip:84783842%40@190.53.38.203>;tag=as4bb74f30
Contact: <sip:101@50.X.X.X:5060>
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" <sip:101@50.X.X.X>;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" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;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" <sip:101@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;tag=as30070ac7
Call-ID: 7f55e32e-e4c6e11a@172.16.8.179
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:101@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@50.X.X.X>;tag=35721c1e3f767ceao4
To: <sip:984783842@50.X.X.X>;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 <xserverlinux@gmail.com>:
--
rickygm
http://gnuforever.homelinux.com
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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asterisk_list at earth... Guest
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Posted: Tue Jan 20, 2015 4:40 am Post subject: [asterisk-users] SEMI-OFFTOPIC openvox |
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On Monday 19 Jan 2015, ricky gutierrez wrote:
Quote: | 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
|
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 .
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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