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jasch

Caller-ID not working anymore

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:P I wish we had wireshark for FXO. The only thing we can do is try to nail the problem down with a couple of gateway builds that give us more logging information (will probably take a few loops).

 

No problem. I am used to beta test stuff.

 

If you can, please load a special build from http://pbxnsip.com/protect/sipfxo-cid-1, and load it into the /pbx directory of the PBX (e.g. using psftp.exe), move the old sipfxo away (e.g. mv sipfxo sipfxo.old.1), then rename sipfxo-cid-1 to sipfxo and restart the box. There should be more logging available that tells us where it sets the caller-ID to Anonymous.

 

I have no idea what psftp is (I'm on MAC) so I replaced sipfxo with new file using SFTP (renamed the previous one to sipfxo.old previously). Restarted the box, and I lost all FXO functionality. I was unable to make any calls, and calls I made to my phone lines wen't unanswered.

 

Any ideas as to what I did wrong?

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No problem. I am used to beta test stuff.

 

 

 

I have no idea what psftp is (I'm on MAC) so I replaced sipfxo with new file using SFTP (renamed the previous one to sipfxo.old previously). Restarted the box, and I lost all FXO functionality. I was unable to make any calls, and calls I made to my phone lines wen't unanswered.

 

Any ideas as to what I did wrong?

 

Sorry, forgot to mention that: check if the sipfxo executable has the "x" permission. Use:

 

cd /pbx
chmod a+rx sipfxo

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Now we're cooking. But still no (useful) PSTN info.

 

[1] 2009/05/15 14:06:59:	Starting up version 3.3.2.3181
[7] 2009/05/15 14:07:03:	Found time zones HST AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+2 GMT+3 GMT+4 GMT+5 GMT+6 GMT+7 GMT+8 GMT+9 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 GMT
[8] 2009/05/15 14:07:03:	Scheduler precision is 10000 us
[1] 2009/05/15 14:07:03:	Working Directory is /pbx
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0
[5] 2009/05/15 14:07:05:	Starting threads
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0:5060
[8] 2009/05/15 14:07:05:	Joined multicast group 224.0.1.75
[7] 2009/05/15 14:07:05:	TCP: Opening socket on 0.0.0.0:5060
[7] 2009/05/15 14:07:05:	TCP: Opening socket on 0.0.0.0:5061
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0
[7] 2009/05/15 14:07:05:	TCP: Opening socket on 0.0.0.0:80
[7] 2009/05/15 14:07:05:	TCP: Opening socket on 0.0.0.0:443
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0:161
[7] 2009/05/15 14:07:05:	UDP: Opening socket on 0.0.0.0:69
[3] 2009/05/15 14:07:41:	PSTN: Channel 0 going to RING
[3] 2009/05/15 14:07:45:	PSTN: Channel 0 going to NO_RING
[3] 2009/05/15 14:07:47:	PSTN: Channel 0 going to RING
[5] 2009/05/15 14:07:47:	PSTN: Did not receive Caller-ID
[9] 2009/05/15 14:07:47:	UDP: Opening socket on 0.0.0.0:16474
[9] 2009/05/15 14:07:47:	UDP: Opening socket on 0.0.0.0:16475
[5] 2009/05/15 14:07:47:	PSTN: Response code: 100
[9] 2009/05/15 14:07:47:	UDP: Opening socket on 0.0.0.0:16410
[9] 2009/05/15 14:07:47:	UDP: Opening socket on 0.0.0.0:16411
[5] 2009/05/15 14:07:48:	PSTN: Response code: 183
[5] 2009/05/15 14:07:51:	Last message repeated 3 times
[3] 2009/05/15 14:07:51:	PSTN: Channel 0 going to NO_RING
[5] 2009/05/15 14:07:51:	PSTN: Response code: 183
[5] 2009/05/15 14:07:57:	Last message repeated 2 times
[5] 2009/05/15 14:07:57:	PSTN: Timeout without ring on 0, going to idle
[3] 2009/05/15 14:07:57:	PSTN: Channel 0 going to IDLE
[5] 2009/05/15 14:07:57:	PSTN: Response code: 200
[5] 2009/05/15 14:07:57:	PSTN: Response code: 487

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[3] 2009/05/15 14:07:41:	PSTN: Channel 0 going to RING
[3] 2009/05/15 14:07:45:	PSTN: Channel 0 going to NO_RING
[3] 2009/05/15 14:07:47:	PSTN: Channel 0 going to RING
[5] 2009/05/15 14:07:47:	PSTN: Did not receive Caller-ID

 

That's the interesting part.

 

So the FXO gateway has the simple impression the line starts ringing, then it stops ringing and starts ringing again. The caller-ID is transported between the NO_RING and the RING. There is simply nothing, no caller-ID.

 

I guess in order to trouble shoot this problem we need to amplify the line or get some measurement.

 

We already had cases there the line volume was too low, and regular PSTN phones were able to read the caller-ID. Maybe we have the same situation here as well. There are cheap line amplifiers available, that's why that would be my first choice.

 

The second option is to employ a tool that measures the line. I am not sure if a digital oscilloscope can do to that job. But we need to see what is going on on the line between those two rings.

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That's the interesting part.

 

So the FXO gateway has the simple impression the line starts ringing, then it stops ringing and starts ringing again. The caller-ID is transported between the NO_RING and the RING. There is simply nothing, no caller-ID.

 

I guess in order to trouble shoot this problem we need to amplify the line or get some measurement.

 

We already had cases there the line volume was too low, and regular PSTN phones were able to read the caller-ID. Maybe we have the same situation here as well. There are cheap line amplifiers available, that's why that would be my first choice.

 

The second option is to employ a tool that measures the line. I am not sure if a digital oscilloscope can do to that job. But we need to see what is going on on the line between those two rings.

 

I still think it's strange that the current line volume is enough for data calls and for every other device I plug in, and it's not for the cs410. I will try to amplify the line, but I just think amplifying it would distort the conversations, since the volume as it is is loud enough without amplifying.

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I still think it's strange that the current line volume is enough for data calls and for every other device I plug in, and it's not for the cs410. I will try to amplify the line, but I just think amplifying it would distort the conversations, since the volume as it is is loud enough without amplifying.

 

I agree. The CS410 is a little bit picky when it comes to signal quality. Cheap phones seem to be better. We had a case where the amplifier really made a difference.

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I agree. The CS410 is a little bit picky when it comes to signal quality. Cheap phones seem to be better. We had a case where the amplifier really made a difference.

 

Did anybody ever resolved this issue??? I'm having the same proble with CS410 version 4.0.1.3499 (Linux), the Caller-ID is not being detected and if I plug a normal phone, the ID shows on the phone LCD. I've already adjusted Input Gain (FXO to IP).

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Did anybody ever resolved this issue??? I'm having the same proble with CS410 version 4.0.1.3499 (Linux), the Caller-ID is not being detected and if I plug a normal phone, the ID shows on the phone LCD. I've already adjusted Input Gain (FXO to IP).

 

I am the original poster. I actually got it resolved two days ago. I had configured Input and Output gain on the PSTN (+4db each). I removed the gain, and everything started working again.

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