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- Jan 2, 2007
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Here is an interesting one...
I have two numbers/phones with Vonage. One connects to my ATA, the other is a softphone number that I use with 3CX.
If I call the softphone number direclty, everything works fine.
If using the Vonage SimulRing feature (rings multiple numbers at the same time) and call my other number (one connected to my ATA), 3CX rejects the call.
Is there a workaround?
Thanks,
Larry
14:07:49.500|DialogUsageManager.cxx(1190)|Trace5|Resip|>>: Got: SipResp: 200 tid=1705ba223e755823 cseq=REGISTER [email protected]:5060 / 5649 from(wire)
14:07:49.500|ClientRegistration.cxx(327)|Trace5|Resip|>>: Clearing service route ([])
14:07:52.921|.\Authorization.cpp(94)|Message5|Authorization|AuthMgr::handle: handle:
INVITE sip:[email protected]:61511 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.27.11:5060;branch=z9hG4bK710E95D360
Max-Forwards: 12
Contact: <sip:216.115.20.41:5060>
To: <sip:[email protected]>
From: <sip:216.115.27.11>;tag=1448886004
Call-ID: [email protected]
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 361
v=0
o=CiscoSystemsSIP-GW-UserAgent 8689 8573 IN IP4 216.115.27.26
s=SIP Call
c=IN IP4 216.115.27.26
t=0 0
m=audio 13830 RTP/AVP 0 18 2 100 101
c=IN IP4 216.115.27.26
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
14:07:52.953|DialogUsageManager.cxx(1190)|Trace5|Resip|>>: Got: SipReq: INVITE [email protected]:61511 tid=23480b29c0db82b48b4153e49814d7f5 cseq=INVITE contact=216.115.20.41:5060 / 101 from(wire)
14:07:52.953|.\CallMgr.cpp(210)|Trace5|CallControl|ADSFactory::createAppDialogSet: Creating ADS for INVITE
14:07:52.953|.\CallConf.cpp(339)|Trace5|CallControl|CallConf::CallConf: Call created: C:3A
14:07:52.953|.\CallConf.cpp(283)|Trace5|CallControl|CallConf::addCallLeg: Added leg# 1 to call C:3A
14:07:52.953|.\Call.cpp(680)|Trace5|CallControl|CallLeg::CallLeg: Leg @L:1@C:3a is created
14:07:52.953|InviteSession.cxx(2046)|Trace5|Resip|>>: Transition UAS_Start -> UAS_Offer
14:07:52.953|.\ISHandler.cpp(21)|Trace5|CallControl|ISHandler:nNewSession: Incoming: sis=14518;oat=Offer;rl=INVITE sip:[email protected]:61511 SIP/2.0
14:07:52.984|.\Call.cpp(38)|Trace5|CallControl|CallLeg:nIncoming: Incoming call is rejected: caller is forbidden
14:07:52.984|.\CallConf.cpp(259)|Log2|CallControl|CallConf::Rejected: Call (C:3A) is rejected: Caller is not allowed
14:07:52.984|ServerInviteSession.cxx(384)|Trace5|Resip|>>: UAS_Offer: reject(403)
14:07:52.984|.\ISHandler.cpp(139)|Trace5|CallControl|ISHandler:nReadyToSend: InviteSession(14518) sends SIP/2.0 403 Forbidden
14:07:52.984|.\ISHandler.cpp(140)|Message5|CallControl|ISHandler:nReadyToSend: IS(14518) sends:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.27.11:5060;branch=z9hG4bK710E95D360
To: <sip:[email protected]>;tag=6a64e15e
From: <sip:216.115.27.11>;tag=1448886004
Call-ID: [email protected]
CSeq: 101 INVITE
Warning: 403 "Caller is not allowed"
Content-Length: 0
I have two numbers/phones with Vonage. One connects to my ATA, the other is a softphone number that I use with 3CX.
If I call the softphone number direclty, everything works fine.
If using the Vonage SimulRing feature (rings multiple numbers at the same time) and call my other number (one connected to my ATA), 3CX rejects the call.
Is there a workaround?
Thanks,
Larry
14:07:49.500|DialogUsageManager.cxx(1190)|Trace5|Resip|>>: Got: SipResp: 200 tid=1705ba223e755823 cseq=REGISTER [email protected]:5060 / 5649 from(wire)
14:07:49.500|ClientRegistration.cxx(327)|Trace5|Resip|>>: Clearing service route ([])
14:07:52.921|.\Authorization.cpp(94)|Message5|Authorization|AuthMgr::handle: handle:
INVITE sip:[email protected]:61511 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.27.11:5060;branch=z9hG4bK710E95D360
Max-Forwards: 12
Contact: <sip:216.115.20.41:5060>
To: <sip:[email protected]>
From: <sip:216.115.27.11>;tag=1448886004
Call-ID: [email protected]
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 361
v=0
o=CiscoSystemsSIP-GW-UserAgent 8689 8573 IN IP4 216.115.27.26
s=SIP Call
c=IN IP4 216.115.27.26
t=0 0
m=audio 13830 RTP/AVP 0 18 2 100 101
c=IN IP4 216.115.27.26
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
14:07:52.953|DialogUsageManager.cxx(1190)|Trace5|Resip|>>: Got: SipReq: INVITE [email protected]:61511 tid=23480b29c0db82b48b4153e49814d7f5 cseq=INVITE contact=216.115.20.41:5060 / 101 from(wire)
14:07:52.953|.\CallMgr.cpp(210)|Trace5|CallControl|ADSFactory::createAppDialogSet: Creating ADS for INVITE
14:07:52.953|.\CallConf.cpp(339)|Trace5|CallControl|CallConf::CallConf: Call created: C:3A
14:07:52.953|.\CallConf.cpp(283)|Trace5|CallControl|CallConf::addCallLeg: Added leg# 1 to call C:3A
14:07:52.953|.\Call.cpp(680)|Trace5|CallControl|CallLeg::CallLeg: Leg @L:1@C:3a is created
14:07:52.953|InviteSession.cxx(2046)|Trace5|Resip|>>: Transition UAS_Start -> UAS_Offer
14:07:52.953|.\ISHandler.cpp(21)|Trace5|CallControl|ISHandler:nNewSession: Incoming: sis=14518;oat=Offer;rl=INVITE sip:[email protected]:61511 SIP/2.0
14:07:52.984|.\Call.cpp(38)|Trace5|CallControl|CallLeg:nIncoming: Incoming call is rejected: caller is forbidden
14:07:52.984|.\CallConf.cpp(259)|Log2|CallControl|CallConf::Rejected: Call (C:3A) is rejected: Caller is not allowed
14:07:52.984|ServerInviteSession.cxx(384)|Trace5|Resip|>>: UAS_Offer: reject(403)
14:07:52.984|.\ISHandler.cpp(139)|Trace5|CallControl|ISHandler:nReadyToSend: InviteSession(14518) sends SIP/2.0 403 Forbidden
14:07:52.984|.\ISHandler.cpp(140)|Message5|CallControl|ISHandler:nReadyToSend: IS(14518) sends:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.20.29:5060
Via: SIP/2.0/UDP 216.115.27.11:5060;branch=z9hG4bK710E95D360
To: <sip:[email protected]>;tag=6a64e15e
From: <sip:216.115.27.11>;tag=1448886004
Call-ID: [email protected]
CSeq: 101 INVITE
Warning: 403 "Caller is not allowed"
Content-Length: 0