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- Mar 6, 2009
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We have used the 3CX PBX successfully with a Linksys SPA-3102 for a few months now. For one reason and another we have decided to move to a Grandstream GWX-4004. I am making very little progress on the configuration of this - would be grateful for some advice.
I have followed all the steps that the 3cx configuration guide suggests.
In 3cx I have added a PSTN line for each of the ports on the gateway (I'll only actually use one). I have added these using the generic template. I am fully up to date with the 3cx software versions.
When I look at the grandstream I can see that all ports have registered successfully. I have installed the latest firmware, versions are:
Program-- 1.0.1.21 Bootloader-- 1.0.0.7 Core-- 1.0.0.25 Base-- 1.0.0.76
When I try an incoming call I get the number not recognised tone. If I disconnect the phone line and redial, I will get the ringing tone.
When I try an outgoing call (over PSTN) I get the ringing tone on the handset, but the dialled number will not ring.
I am assuming this something to do with my configuration of the grandstream, probably in the "call progress tones" section(?). I have tried various settings here that I have found on the web including the ones shown in the attached screenshot. I tried these with and without the ch1-4: prefix.
Any help much appreciated
Ian
========================================================
Server status log for outgoing call is below:
14:51:52.491 Call::Terminate [CM503008]: Call(26): Call is terminated
14:51:26.303 Line:
rintEndpointInfo [CM505002]: Gateway:[GRANDSTREAM] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW-4004 V1.3A 1.0.1.21] Transport: [sip:192.168.176.115:5060]
14:51:26.303 CallCtrl:
nAnsweredCall [CM503002]: Call(26): Alerting sip:[email protected]:5060
14:51:26.178 MediaServerReporting::SetRemoteParty [MS210002] C:26.2:Offer provided. Connection(transcoding mode): 192.168.176.115:7448(7449)
14:51:26.116 MediaServerReporting::SetRemoteParty [MS210000] C:26.1:Offer received. RTP connection: 192.168.176.189:10142(10143)
14:51:26.116 CallCtrl:
nSelectRouteReq [CM503004]: Call(26): Calling: PSTNline:10002@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]
14:51:26.053 CallCtrl:
nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected];user=phone]
14:51:26.053 CallLeg::setRemoteSdp Remote SDP is set for legC:26.1
14:51:26.053 Extension:
rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [elmegIP290-3.4] Transport: [sip:192.168.176.115:5060]
14:51:26.037 CallCtrl:
nIncomingCall [CM503001]: Call(26): Incoming call from Ext.101 to [sip:[email protected];user=phone]
14:51:25.975 CallLeg:
nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.176.189:5060;branch=z9hG4bK-1hwo6nshorgy;rport=5060
Max-Forwards: 70
Contact: [sip:[email protected]:5060;line=ga88nc7d]
To: [sip:[email protected];user=phone]
From: "101"[sip:[email protected]];tag=n3ttyqckgp
Call-ID: 3c54dd35ef42-7zcumpn47kb1@192-168-176-189
CSeq: 2 INVITE
Session-Expires: 3600
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12881170285:3695341f8d6448b1fe0e97498f65ade9",uri="sip:[email protected];user=phone",response="9385fbee054e74f93393b75ec9f97fdd",algorithm=md5
Supported: timer, 100rel, replaces
User-Agent: elmegIP290-3.4
Allow-Events: talk, hold, refer
Content-Length: 0
P-Key-Flags: keys="3"
=====================================
=====================================
=====================================
I have attached the settings I currently have for the tones
I have followed all the steps that the 3cx configuration guide suggests.
In 3cx I have added a PSTN line for each of the ports on the gateway (I'll only actually use one). I have added these using the generic template. I am fully up to date with the 3cx software versions.
When I look at the grandstream I can see that all ports have registered successfully. I have installed the latest firmware, versions are:
Program-- 1.0.1.21 Bootloader-- 1.0.0.7 Core-- 1.0.0.25 Base-- 1.0.0.76
When I try an incoming call I get the number not recognised tone. If I disconnect the phone line and redial, I will get the ringing tone.
When I try an outgoing call (over PSTN) I get the ringing tone on the handset, but the dialled number will not ring.
I am assuming this something to do with my configuration of the grandstream, probably in the "call progress tones" section(?). I have tried various settings here that I have found on the web including the ones shown in the attached screenshot. I tried these with and without the ch1-4: prefix.
Any help much appreciated
Ian
========================================================
Server status log for outgoing call is below:
14:51:52.491 Call::Terminate [CM503008]: Call(26): Call is terminated
14:51:26.303 Line:
14:51:26.303 CallCtrl:
14:51:26.178 MediaServerReporting::SetRemoteParty [MS210002] C:26.2:Offer provided. Connection(transcoding mode): 192.168.176.115:7448(7449)
14:51:26.116 MediaServerReporting::SetRemoteParty [MS210000] C:26.1:Offer received. RTP connection: 192.168.176.189:10142(10143)
14:51:26.116 CallCtrl:
14:51:26.053 CallCtrl:
14:51:26.053 CallLeg::setRemoteSdp Remote SDP is set for legC:26.1
14:51:26.053 Extension:
14:51:26.037 CallCtrl:
14:51:25.975 CallLeg:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.176.189:5060;branch=z9hG4bK-1hwo6nshorgy;rport=5060
Max-Forwards: 70
Contact: [sip:[email protected]:5060;line=ga88nc7d]
To: [sip:[email protected];user=phone]
From: "101"[sip:[email protected]];tag=n3ttyqckgp
Call-ID: 3c54dd35ef42-7zcumpn47kb1@192-168-176-189
CSeq: 2 INVITE
Session-Expires: 3600
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12881170285:3695341f8d6448b1fe0e97498f65ade9",uri="sip:[email protected];user=phone",response="9385fbee054e74f93393b75ec9f97fdd",algorithm=md5
Supported: timer, 100rel, replaces
User-Agent: elmegIP290-3.4
Allow-Events: talk, hold, refer
Content-Length: 0
P-Key-Flags: keys="3"
=====================================
=====================================
=====================================
I have attached the settings I currently have for the tones