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joe

MOH (registeres user)

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Hi,

 

THe MOH feature isn't working for me. I get a 'not found' response from PBX.

I don't know much about PBXNSIP so could someone give me some simple steps as to how i can get

this feature to work please.

 

I using a VOIP gateway to which i have analog phones plugged in.

All i have told the PBX is to play MOH. Can i assume that MOH is played when a SIP message with no SDP is sent to the PBX?

 

Regards,

Joe

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hi,

everything is as per default.

MOH.wav file is selected.

what triggers the PBX to play music?

Nobody has been able to clarify this for me

 

Cheers

 

 

hi,

 

When you first install pbxnsip with the standard config it should just work.

Did you change the moh sound source?

 

tx

matt

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hi,

everything is as per default.

MOH.wav file is selected.

what triggers the PBX to play music?

Nobody has been able to clarify this for me

 

Cheers

 

Please verify whether you have a folder 'audio_moh' under the working directory. If not, please download it from the pbxnsip website.

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what OS?

 

if win2008 make sure the sound is enabled..(whatever it is called)

 

I suggest running this all on win xp pro the first time to avoid all that stuff for testing...getting yourself up to speedd...

 

i have a $99 getting started with pbxnsip ebook if you like...

 

matt

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Win Xp Pro.

 

If i use an an SJ Phone to put the other party on hold then that works fine.

 

But using a Voip Gateway does not. The question is whether PBXNSIP supports

draft-ietf-sipping-service-examples-11.

 

I noticed that the SJ Phone sends re-invites to the PBX but the sip messaging does not follow the draft mentioned above.

 

Thanks

 

 

what OS?

 

if win2008 make sure the sound is enabled..(whatever it is called)

 

I suggest running this all on win xp pro the first time to avoid all that stuff for testing...getting yourself up to speedd...

 

i have a $99 getting started with pbxnsip ebook if you like...

 

matt

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Win Xp Pro.

 

If i use an an SJ Phone to put the other party on hold then that works fine.

 

But using a Voip Gateway does not. The question is whether PBXNSIP supports

draft-ietf-sipping-service-examples-11.

 

I noticed that the SJ Phone sends re-invites to the PBX but the sip messaging does not follow the draft mentioned above.

 

Thanks

 

Are you talking about the registered users to the PBX or peer-to-peer interaction?

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Registered users to PBX.

 

I looked at the log file and just before the PBX informed me of 'Not Found', it seemed to look

in the dial plan. Unfortunately i don't know how to set a dial for the pbx. I can't find any documentation.

 

Cheers

 

Are you talking about the registered users to the PBX or peer-to-peer interaction?

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Registered users to PBX.

 

I looked at the log file and just before the PBX informed me of 'Not Found', it seemed to look

in the dial plan. Unfortunately i don't know how to set a dial for the pbx. I can't find any documentation.

 

Cheers

 

Not sure what you are referring to. At this moment it is better if you can send email to support@pbxnsip.com with the SIP logging enabled. Please see on how to get the SIP logging. https://www.pbxnsipsupport.com/index.php?_m...;kbarticleid=58

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