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Unread 07-28-2009, 08:39 AM   #1
richard99
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Default Voxalot failing call Pbxes Ext.

Using for quite long time to call Pbxes extension, but since few day not able to call pbxes extension form Voxalot speed dial.

I remember there is was same issue in past but that time due to one of US server was down. But now all server are available. Any comment from other users facing same problem to call Pbxes ext.
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Unread 07-28-2009, 09:58 AM   #2
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Originally Posted by richard99 View Post
Using for quite long time to call Pbxes extension, but since few day not able to call pbxes extension form Voxalot speed dial.

I remember there is was same issue in past but that time due to one of US server was down. But now all server are available. Any comment from other users facing same problem to call Pbxes ext.
Is it just PBXes from the speed dial you are having problems with?
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Unread 07-28-2009, 10:02 AM   #3
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Yes only pbxes from speed dial other i.e callcentric,poivy,voipbuster and other all working fine
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Unread 07-28-2009, 10:12 AM   #4
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Yes only pbxes from speed dial other i.e callcentric,poivy,voipbuster and other all working fine
Can you call using dial plan and not speed dial?
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Unread 07-28-2009, 10:30 AM   #5
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No. I can not call using dial plan any pbxes extension same ...busy tone. all other provider working normal using dial plan.

Last edited by richard99; 07-28-2009 at 10:36 AM.
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Unread 07-28-2009, 02:03 PM   #6
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It has always been a bit problematic reaching a PBXes extension from outside PBXes...

What SIP URI format were you using ... and anything setup at PBXes like a trunk carrying the extension name to make this possible... ?

You see the problem here is that if everything else is working except PBXes the chances are quite good that this is more likely a PBXes issue than a VoXalot issue...
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Unread 07-28-2009, 02:29 PM   #7
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I'm not sure if it's worth mentioning, but I got an email this morning from PBXes that they had a service outage on one of the servers that I was hosted on. Paying subscribers were moved over to other servers. Free subscribers had to wait until the outage was resolved. For free subscribers, you do have to re-enable your trunks once the outage is resolved.
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Unread 07-28-2009, 04:19 PM   #8
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I am paying subscriber of Pbxes, using <account name>-<extension no.> via Pbxes and Pbxes is defined in Voice service provider. If only extension is made without defining in inbound routing , calling from out side definitely will get busy tone.

Even tried as sip URL like sip:<account name>-<extension no.>@pbxes.org but same busy tone.
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Unread 07-28-2009, 05:07 PM   #9
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Originally Posted by richard99 View Post
I am paying subscriber of Pbxes, using <account name>-<extension no.> via Pbxes and Pbxes is defined in Voice service provider. If only extension is made without defining in inbound routing , calling from out side definitely will get busy tone.

Even tried as sip URL like sip:<account name>-<extension no.>@pbxes.org but same busy tone.
Ok...

-So you have PBXes Extension 1 setup as a provider in VoXalot
-You're trying to have VoXalot use PBXes Extension 1 to call PBXes Extension 2...is that about right?
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Unread 07-28-2009, 05:20 PM   #10
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Yes, I am using this setup for quite long time. Following is log..
Vox281 is id of Voxalot xxxxxx and calling **700 where pbxes is defined...the problem seems in following highlighted sip response..

VERBOSE[17935] logger.c: -- Called Vox281/**700
VERBOSE[13298] chan_sip.c: SIP response 407 to standard invite
VERBOSE[13298] logger.c: We're at 78.46.102.143 port 40078
VERBOSE[13298] logger.c: Video is at 78.46.102.143 port 41708
VERBOSE[13298] logger.c: Adding codec 0x4 (ulaw) to SDP
VERBOSE[13298] logger.c: Adding codec 0x8 (alaw) to SDP
VERBOSE[13298] logger.c: Adding codec 0x10 (g726) to SDP
VERBOSE[13298] logger.c: Adding codec 0x400 (ilbc) to SDP
VERBOSE[13298] logger.c: Adding codec 0x2 (gsm) to SDP
VERBOSE[13298] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
VERBOSE[13298] chan_sip.c: SIP response 100 to standard invite
VERBOSE[13298] chan_sip.c: SIP response 100 to standard invite
VERBOSE[13298] logger.c: -- Got SIP response 603 "Declined"
VERBOSE[17935] logger.c: -- SIP/Vox281-f577 is busy
VERBOSE[17935] chan_sip.c: Hangup call SIP/Vox281-f577, SIP callid 701d75b307ea847f51f740943c45ad5e@us.voxalot.com
VERBOSE[17935] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
VERBOSE[17935] chan_sip.c: Hangup call SIP, SIP callid admt2zmgi665.1qcq35q8x1eyitg@188.40.65.148
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