Outgoing call report problem Elastix 2.0

maestrin

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#1
I face a strange issue with Elastix 2.0.1 and OpenVox A400E 2FXS,2FXO (dahdi). Icoming and outgoing calls work fine but all outgoing calls reported as answered even the call isnt answered or busy. The billing module starts billing an unanswered call as soon as the phone rings.I have also reported this as a bug. Has anyone any idea how to resolve this?
 

azido

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#2
I got the same problem anybody here have a fix?
 

jgutierrez

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#3
That is due to the fact that when you dial out, asterisk gets "answered" by your dahdi trunks, that is whay you get "answered" state on your calls.
If you are going to use analog lines, you will need to ask your provider to enable "reversal polarity" on your lines.
 

maestrin

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#4
Thanks jgutierrez
I dont have any polarity reversal settings in my chan_dahdi.conf and I dont know if my provider supports it. You mean this?
cidstart = polarity
cidsignalling = dtmf
 

jgutierrez

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#5
I mean on /etc/asterisk/chan_dahdi.conf:
answeronpolarityswitch=yes

But if your provider doesn't support/enable it, it wont work.
 

maestrin

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#6
Probably my provider doesnt support it. I tried callproress=yes and it seems like working. I am not sure if it brings up other problems.
Still testing...
 

maestrin

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#7
I think it has nothing to do with the PSTN trunk. I tested with a sip trunk and i have exactly the same issue.
- Executing [s@macro-dialout-trunk:19] Dial("SIP/550-0000001f", "SIP/MAIN/2xxxxxxxxx,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called MAIN/2xxxxxxxxx
-- SIP/MAIN-00000020 is ringing
-- SIP/MAIN-00000020 answered SIP/550-0000001f
-- Packet2Packet bridging SIP/550-0000001f and SIP/MAIN-00000020....

the call isnt answered. any idea?
 

dicko

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#8
callproress=yes

or I assume a typo and you actually used:-

callprogress=yes ; :)

works by default only for NANP dial indications and even then is very flaky (at least that was the case last time I tried it)

perhaps :-

http://www.asteriskguru.com/tutorials/r ... email.html


might help understand how DAHDI/(old zaptel) works

If your provider supports it the answeronpolarityswitch=yes should resolve your problem, if not then . . .

http://www.voip-info.org/wiki/view/Aste ... dahdi.conf

is quite authoritative as to what you can do within your limitations

if for the SIP calls you have

canreinvite=yes

all bets are off.
 

maestrin

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#9
Thanks for the infos.
Unfortunately its doesnt work.
It seems like asterisk answers the call before it actually answered.It happens only to outbound calls.
-- SIP/MAIN-0822f3f8 is ringing
-- SIP/MAIN-0822f3f8 answered SIP/550-0822a8b0

By calling to another sip extension it reports fine.
-- Called 101
-- SIP/101-08233758 is ringing
could that be there is something wrong with the macro-dialout-trunk in extensions.conf?
 

dicko

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#10
No, it just works normally.

who or what is your SIP/MAIN endpoint?

asterisk per se is not answering it, your sip/MAIN context is
 

maestrin

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#11
MAIN is the name of the sip trunk
 

dicko

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#12
Then investigate why your sip trunk gives answer immediately before connection, only you and your provider can do that, maybe they are ripping you off ?

if your

http://<server>/admin/reports.php

similarly show connections, then you messed up, if that page disagrees with your call lengths and connections apropos what you post your original concerns about , then I believe you have unearthed yet another Elastix 2.0 bug, if so please file in the bug-tracker below appropriately.

dicko
 

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