keywords: ip pbx voip gateway gsm gateway

× Questions about VS-GW2120/GW1600/GW1202 and WGW1002G GSM Series VoIP Gateway

route call

6 years 10 months ago #10152 by rickygm
I have a gw WGW1002G , I followed the steps in this manual and unable to make or receive calls
www.lojamundi.com.br/download/gateways-g..._Asterisk_Server.pdf

the sip account is registered in my asterisk server!

any idea?
6 years 10 months ago #10153 by lindali
hi,

This is Linda from openvox. I'll help you with your probem. For convenient contact, you can add my skype.

best regards,

skype: linda.li258
email: This email address is being protected from spambots. You need JavaScript enabled to view it.
6 years 10 months ago #10154 by lindali
hi,

This is Linda from openvox. I'll help you with your probem. For convenient contact, you can add my skype.

best regards,

skype: linda.li258
email: This email address is being protected from spambots. You need JavaScript enabled to view it.
6 years 10 months ago #10155 by rickygm

lindali wrote: hi,

This is Linda from openvox. I'll help you with your probem. For convenient contact, you can add my skype.

best regards,

skype: linda.li258
email: This email address is being protected from spambots. You need JavaScript enabled to view it.



you are in my skype, but only see you offline
6 years 10 months ago #10156 by rickygm
HI, yet I can not solve this problem

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

Contact: <sip:[email protected]:5060>
Content-Length: 0


<
>
-- Executing [[email protected]_pstn:1] Dial("SIP/101-0000004e", "SIP/5001/[email protected],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%[email protected] SIP/2.0
Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport
Max-Forwards: 70
From: "Operadora" <sip:[email protected]>;tag=as3708c762
To: <sip:84783842%[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;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/[email protected]

<--- 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:[email protected]>;tag=35721c1e3f767ceao4
To: <sip:[email protected]>;tag=as77fb37e2
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
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:[email protected]: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:[email protected]>;tag=as3708c762
To: <sip:84783842%[email protected]>;tag=as4bb74f30
Call-ID: [email protected]: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%[email protected] SIP/2.0
Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport
Max-Forwards: 70
From: "Operadora" <sip:[email protected]>;tag=as3708c762
To: <sip:84783842%[email protected]>;tag=as4bb74f30
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as3708c762'
Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]_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:[email protected]>;tag=35721c1e3f767ceao4
To: <sip:[email protected]>;tag=as77fb37e2
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
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:[email protected] SIP/2.0
Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36
From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4
To: <sip:[email protected]>;tag=as30070ac7
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
CSeq: 101 ACK
Max-Forwards: 70
Contact: "101" <sip:[email protected]: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:[email protected]>;tag=35721c1e3f767ceao4
To: <sip:[email protected]>;tag=as77fb37e2
Call-ID: This email address is being protected from spambots. You need JavaScript enabled to view it.
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
6 years 10 months ago #10157 by rickygm
the gw is receiving calls, but not routed asterisk


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