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SPA3102 Config help

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xjgper

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Good day folks-

New to the forums, and new to 3cx..

I'm trying to replicate something similar to our mitel pbx w/SIP utilizing the 3cx software and a spa3102. This project is for a very small 2 person office..

Ideally i'm trying to go from PSTN - 3cx - Softphone (and the like)

I'm trying to use a SPA3102... I've got it setup on 3cx and its talking, however i'm sure there is something i'm not doing right to get outbound and inbound on the SPA sent to and from the 3cx extensions..

When trying to dial out from an ext 3cx gives this log..

12:59:21.203 [CM503008]: Call(8): Call is terminated
12:59:21.203 [CM503015]: Call(8): Attempt to reach <sip:[email protected]> failed. Reason: Server Failure
12:59:21.187 [CM503003]: Call(8): Call to sip:[email protected]:5062 has failed; Cause: 503 Service Unavailable; from IP:172.16.10.100:5062
12:59:21.125 [CM503024]: Call(8): Calling PSTNline:7503@(Ln.10000@spa1)@[Dev:sip:[email protected]:5062]
12:59:21.109 [CM503004]: Call(8): Route 1: PSTNline:7503@(Ln.10000@spa1)@[Dev:sip:[email protected]:5062]
12:59:21.109 [CM503010]: Making route(s) to <sip:[email protected]>


I'm asuming the service unavailable error is due to some lack of outbound call rule on the SPA.

also i can not seem to get the spa to pick up, i get a modem type screech when the line is dialed from teh outside pstn

Please give me your input!

Thanks,
Dave
 
So you are trying to dial out "7503" on your PSTN line? Does the main status page show that the 3102 thinks that the PSTN line is idle?

Does the 3102 show as registered on the 3CX? Was it used previously, could there be some old setting left in that need to be changed. You're sure that the modem screech is coming from the 3102, nothing else (Fax machine, computer modem) plugged into the same line? What does an incoming call log look like?

The 3102 has a built in Syslog server that can be set to send logs to another computer running free syslog software elsewhere on your network. This can provide detailed logs that are a great help in narrowing down what the problem might be.
 
The 3102 does show registered on the 3cx. Yes i did play with it previously, modified some settings, but found a page in this forums indicating a factory reset via the phone port of the 3102. I haven't setup the syslog server yet but will do so.

I doesn't matter what i'm trying to dial out it doesnt go, whether it be 4 digits, 7 digits, or a full long distance #.

There is nothing hooked on teh 3102 currently (i removed the handset after i did teh reset factory defaults), i have the "line" port hooked into the ASU of the mitel (analog service unit) to get pots like functionality, hence me dialing the 7503 which is another internal ext in teh building.

I guess i need to figure out why the 3102 is rejecting the request to dial out, and why i can't seem to get it to accept an incoming call with out screeches..

If i replace the config on the 3102 with the one generated by the 3cx will that ensure that any other setting is as to be expected by the 3cx?

Thanks!

Dave
 
A bunch of gobly gook for your phone inclined folk! Thanks!!!


Code: 
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	ACK sip:[email protected]:5062 SIP/2.0<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-;rport<013><010>Max-Forwards: 70<013><010>To: <sip:[email protected]:5062>;tag=165ae75264062440i1<013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 ACK<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(371)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(371)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	SIP/2.0 503 Service Unavailable<013><010>To: <sip:[email protected]:5062>;tag=165ae75264062440i1<013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(374)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(374)
07-15-2009	10:19:44	Local2.Debug	172.16.10.100	AUD:Stop PSTN Tone
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	SIP/2.0 100 Trying<013><010>To: <sip:[email protected]:5062><013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(338)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(338)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	INVITE sip:[email protected]:5062 SIP/2.0<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-;rport<013><010>Max-Forwards: 70<013><010>Contact: <sip:[email protected]:5060><013><010>To: <sip:[email protected]:5062><013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO<013><010>Content-Type: application/sdp<013><010>User-Agent: 3CXPhoneSystem 7.1.7060.0<013><010>Content-Length: 270<013><010><013><010>v=0<013><010>o=3cxPS 467430014976 37849399297 IN IP4 172.16.10.14<013><010>s=3cxPS Audio call<013><010>c=IN IP4 172.16.10.1<013><010>t=0 0<013><010>m=audio 40006 RTP/AVP 0 8 3 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:3 GSM/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=sendrecv
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(820)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(820)
 
xjgper said:
A bunch of gobly gook for your phone inclined folk! Thanks!!!


Code: 
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	ACK sip:[email protected]:5062 SIP/2.0<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-;rport<013><010>Max-Forwards: 70<013><010>To: <sip:[email protected]:5062>;tag=165ae75264062440i1<013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 ACK<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(371)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(371)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	SIP/2.0 503 Service Unavailable<013><010>To: <sip:[email protected]:5062>;tag=165ae75264062440i1<013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(374)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(374)
07-15-2009	10:19:44	Local2.Debug	172.16.10.100	AUD:Stop PSTN Tone
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	SIP/2.0 100 Trying<013><010>To: <sip:[email protected]:5062><013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-<013><010>Server: Linksys/SPA3102-3.3.6(GW)<013><010>Content-Length: 0
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(338)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]->172.16.10.14:5060(338)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local0.Info	172.16.10.100	<010>
07-15-2009	10:19:44	Local7.Debug	172.16.10.100	INVITE sip:[email protected]:5062 SIP/2.0<013><010>Via: SIP/2.0/UDP 172.16.10.14:5060;branch=z9hG4bK-d8754z-cd4fdf38572ac133-1---d8754z-;rport<013><010>Max-Forwards: 70<013><010>Contact: <sip:[email protected]:5060><013><010>To: <sip:[email protected]:5062><013><010>From: "3CX VoIP Phone"<sip:[email protected]:5060>;tag=e17b2e74<013><010>Call-ID: OTYyY2M1OGYyZTc4ZGI4Zjc2Y2RjOGIzZDNjOTc2ODA.<013><010>CSeq: 1 INVITE<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO<013><010>Content-Type: application/sdp<013><010>User-Agent: 3CXPhoneSystem 7.1.7060.0<013><010>Content-Length: 270<013><010><013><010>v=0<013><010>o=3cxPS 467430014976 37849399297 IN IP4 172.16.10.14<013><010>s=3cxPS Audio call<013><010>c=IN IP4 172.16.10.1<013><010>t=0 0<013><010>m=audio 40006 RTP/AVP 0 8 3 101<013><010>a=rtpmap:0 PCMU/8000<013><010>a=rtpmap:8 PCMA/8000<013><010>a=rtpmap:3 GSM/8000<013><010>a=rtpmap:101 telephone-event/8000<013><010>a=fmtp:101 0-15<013><010>a=sendrecv
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(820)
07-15-2009	10:19:44	Local0.Info	172.16.10.100	[1]<<172.16.10.14:5060(820)
Click to expand...
Dave,

Sorry, but this log should be addressed to SPA3102 support...

Thanks
 
Is the Mitel board putting out 24 volts or 48? I had a problem with the on hook voltage setting when i first tried hooking an SPA3000 to my Mitel board years ago. Because the default is set for a 48 volt line the 24 volts caused the 3000 to link the line was off-hook and would not allow a call.
 
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