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- Nov 25, 2015
- Messages
- 244
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Hello all,
There was an analogue phone that needed to be connected to 3CX. I found an Audiocodes MP-112 ATA and connected it to the phone and 3CX.
It is one of the most difficult devices to set-up if you're new to them, so I spent quite some time getting it to register. But it registers as an extension just fine now (ext 15) and I can call the extension and the phone rings and I have two-way audio.
There is, however, one thing that doesn't work and it's outbound calling.
It seems 3CX is unable to match the call to any useraccount.
I'll post the logs below, but if you look at them, you'll see the first request from the ATA to 3CX is without authentication, as it's supposed to be. 3CX then sends a MD5-challenge and the ATA makes it's request again, this time with the answer to that challenge. This is also accepted by 3CX, because when I change the password to something wrong in the ATA, I get a clear 'authentication failed'-error instead.
So, I have tried and changed every setting (there must be over a 1000 in the ATA) that seemed relevant to making calls in the ATA, even tried adding it as a PSTN-gateway, but to no avail. The error is always the same, it seems.
Any help would be greatly appreciated!
03-mrt-2016 01:40:33.457 Leg L:65.1[Unknown:] is terminated: Cause: BYE from PBX
03-mrt-2016 01:40:33.407 [CM502001]: Source info: From: "15"<sip:[email protected]>;tag=1c200100638; To: <sip:[email protected];user=phone>
03-mrt-2016 01:40:33.407 [CM503013]: Call(C:65): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from IP.OF.THE.PHONE:5060 tid=ac200490246 Call-ID=20009976433201614037 @IP.OF.THE.PHONE:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP IP.OF.THE.PHONE;branch=z9hG4bKac200490246
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "15"<sip:[email protected]>;tag=1c200100638
Call-ID: 20009976433201614037 @IP.OF.THE.PHONE
CSeq: 2 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Content-Disposition: session
Content-Type: application/sdp
Proxy-Authorization: Digest username="15",realm="3CXPhoneSystem",nonce="414d535c0ce8190146:5ff9e1a2080835d2039b7e099cd28909",uri="sip:[email protected]",algorithm=MD5,response="d52a7251744e6587c04f1eb3c49d6421"
Supported: em, 100rel, timer, replaces, path, early-session, resource-priority, sdp-anat
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.6.20A.022.003
Content-Length: 252
v=0
o=AudiocodesGW 200097660 200097541 IN IP4 IP.OF.THE.PHONE
s=Phone-Call
c=IN IP4 IP.OF.THE.PHONE
t=0 0
m=audio 6030 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
03-mrt-2016 01:40:33.255 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from IP.OF.THE.PHONE:5060 tid=ac200103763 Call-ID=20009976433201614037 @IP.OF.THE.PHONE:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP IP.OF.THE.PHONE;branch=z9hG4bKac200103763
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "15"<sip:[email protected]>;tag=1c200100638
Call-ID: 20009976433201614037 @IP.OF.THE.PHONE
CSeq: 1 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Content-Disposition: session
Content-Type: application/sdp
Supported: em, 100rel, timer, replaces, path, early-session, resource-priority, sdp-anat
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.6.20A.022.003
Content-Length: 252
v=0
o=AudiocodesGW 200097660 200097541 IN IP4 IP.OF.THE.PHONE
s=Phone-Call
c=IN IP4 IP.OF.THE.PHONE
t=0 0
m=audio 6030 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
There was an analogue phone that needed to be connected to 3CX. I found an Audiocodes MP-112 ATA and connected it to the phone and 3CX.
It is one of the most difficult devices to set-up if you're new to them, so I spent quite some time getting it to register. But it registers as an extension just fine now (ext 15) and I can call the extension and the phone rings and I have two-way audio.
There is, however, one thing that doesn't work and it's outbound calling.
It seems 3CX is unable to match the call to any useraccount.
I'll post the logs below, but if you look at them, you'll see the first request from the ATA to 3CX is without authentication, as it's supposed to be. 3CX then sends a MD5-challenge and the ATA makes it's request again, this time with the answer to that challenge. This is also accepted by 3CX, because when I change the password to something wrong in the ATA, I get a clear 'authentication failed'-error instead.
So, I have tried and changed every setting (there must be over a 1000 in the ATA) that seemed relevant to making calls in the ATA, even tried adding it as a PSTN-gateway, but to no avail. The error is always the same, it seems.
Any help would be greatly appreciated!
03-mrt-2016 01:40:33.457 Leg L:65.1[Unknown:] is terminated: Cause: BYE from PBX
03-mrt-2016 01:40:33.407 [CM502001]: Source info: From: "15"<sip:[email protected]>;tag=1c200100638; To: <sip:[email protected];user=phone>
03-mrt-2016 01:40:33.407 [CM503013]: Call(C:65): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from IP.OF.THE.PHONE:5060 tid=ac200490246 Call-ID=20009976433201614037 @IP.OF.THE.PHONE:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP IP.OF.THE.PHONE;branch=z9hG4bKac200490246
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "15"<sip:[email protected]>;tag=1c200100638
Call-ID: 20009976433201614037 @IP.OF.THE.PHONE
CSeq: 2 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Content-Disposition: session
Content-Type: application/sdp
Proxy-Authorization: Digest username="15",realm="3CXPhoneSystem",nonce="414d535c0ce8190146:5ff9e1a2080835d2039b7e099cd28909",uri="sip:[email protected]",algorithm=MD5,response="d52a7251744e6587c04f1eb3c49d6421"
Supported: em, 100rel, timer, replaces, path, early-session, resource-priority, sdp-anat
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.6.20A.022.003
Content-Length: 252
v=0
o=AudiocodesGW 200097660 200097541 IN IP4 IP.OF.THE.PHONE
s=Phone-Call
c=IN IP4 IP.OF.THE.PHONE
t=0 0
m=audio 6030 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
03-mrt-2016 01:40:33.255 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from IP.OF.THE.PHONE:5060 tid=ac200103763 Call-ID=20009976433201614037 @IP.OF.THE.PHONE:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP IP.OF.THE.PHONE;branch=z9hG4bKac200103763
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "15"<sip:[email protected]>;tag=1c200100638
Call-ID: 20009976433201614037 @IP.OF.THE.PHONE
CSeq: 1 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Content-Disposition: session
Content-Type: application/sdp
Supported: em, 100rel, timer, replaces, path, early-session, resource-priority, sdp-anat
User-Agent: Audiocodes-Sip-Gateway-MP-112 FXS/v.6.20A.022.003
Content-Length: 252
v=0
o=AudiocodesGW 200097660 200097541 IN IP4 IP.OF.THE.PHONE
s=Phone-Call
c=IN IP4 IP.OF.THE.PHONE
t=0 0
m=audio 6030 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv