Incoming calls from PSTN or MS Lync always ends with 486 Err

1. Brekeke Product Name and version:
Brekeke PBX, Version 2.4.9.0 , Basic
2. Java version:
Java6 Update31
3. OS type and the version:
Windows Server 2008 R2 SP1 Enterprise
4. UA (phone), gateway or other hardware/software involved:
lync, x-lite, pstn
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
Patern 9
6. Your problem:
Incoming calls from PSTN or MS Lync always ends with Busy/Error 486 (in voicemail).
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I have a problem with calling on numbers which ends in Brekeke PBX. It looks like that ARS rule is working correctly, because I can see dial numbers correctly registered in SIP server by PBX. Also ARS rule use configured PBX extension/user because it is changing behavior depending on voicemail settings under PBX extension account.
I want to have incoming calls redirected to local SIP account (to the SIP phones registered in the PBX' SIP server). For forwarding destinations field Im using syntax like "sip:sipaccount@". With voicemail checked I always hear the voicemail when calling to that number. In the PBX' sv log I see result=Busy error=486 on these calls.

My ARS rule is simple, it has only one IN pattern with
Matching
To: sip:&v1@

Deploy
To: &v3

Thanks in advance.

Interesting thing is, that it partially works if Im calling from registered UA. The problem in this case is that I can hear voice only in one direction.

Please suggest what can be wrong.

Do I need patterns OUT in the ARS rule when Im receiving calls from registered UA or from external PSTN???

Interesting thing is, that it partially works if Im calling from registered UA. The problem in this case is that I can hear voice only in one direction.

if call directly between these two registered UA, is there one-way voice problem?

I want to have incoming calls redirected to local SIP account (to the SIP phones registered in the PBX' SIP server).

Do you want to bypass pbx for calls from pstn or MS Lync?

if call directly between these two registered UA, is there one-way voice problem?

There is no problem between two registered UA (for example when calling other sip name using x-lite).

Do you want to bypass pbx for calls from pstn or MS Lync?

Yes, also. I want to achieve several thinks:
a) be able to call PBX users from Lync
b) be able to pass calls from Lync to PSTN
c) be able to receive calls from PSTN by PBX users
d) be able to receive calls from PSTN by Lync users

ad a) this is working when directly routed using Dial plan; not working when using ARS (486 busy error)

ad b) this is not working for me; request is passed to PSTN, but PSTN returns SIP packets which PBX is unable to parse (com.brekeke.net.sip.InvalidSIPException: No headers)

ad c) this is working when directly routed using Dial plan; not working when using ARS

ad d) i will try after resolving c)

I have made little progress:
ad b) this is working now using PBX ARS, where I had to configure both IN and OUT patterns for calling to PSTN. Still I don't understand, why it doesnt work using Dial plan only (I dont know but it can be related to specific requirements of PSTN).

I will be continuing get other tasks working.

I'm not sure i understand your problem, but i think you are saying you get it to work with the sip server only but when using the ARS you are having problems.

My understanding is you can not direct a call from the ARS to a sip address outside the pbx. You can take an incoming call and send it to an extension that is registered (or to vm), but you can't put the destination sip:xxx.xxx.xxx.xxx:5060. I believe if you want to do that you need to send the call to an extension where you then can call forward it to a sip address. Older versions of Brekeke used to allow this but for security it was changed to require passing thru an extension/user. You can however use a telephone number as the destination in the ARS that will get processed by another entry in the ARS for say call forwarding a DID to another DID. Hope this helps.

voipwell.com wrote:I'm not sure i understand your problem, but i think you are saying you get it to work with the sip server only but when using the ARS you are having problems.

My understanding is you can not direct a call from the ARS to a sip address outside the pbx. You can take an incoming call and send it to an extension that is registered (or to vm), but you can't put the destination sip:xxx.xxx.xxx.xxx:5060. I believe if you want to do that you need to send the call to an extension where you then can call forward it to a sip address. Older versions of Brekeke used to allow this but for security it was changed to require passing thru an extension/user. You can however use a telephone number as the destination in the ARS that will get processed by another entry in the ARS for say call forwarding a DID to another DID. Hope this helps.

Thanks for the info, I didn't know that. That helped me to move forward.
Currently I have only one problem:
Every call which is routed through ARS (from PSTN to SIP or oposite way) has following behavior:
1) call is terminated after 15 secs (approx)
2) i can hear only voice only in one direction

I believe that #1 and #2 are caused by same misconfiguration.
Any suggestions?

Many thanks.

Im closing this "all-problems-in-one" topic and opening new one.