Voxalot failing call Pbxes Ext.

richard99 · 07-28-2009, 08:39 AM

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.

martin · 07-28-2009, 09:58 AM

Quote:

Originally Posted by richard99

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?

richard99 · 07-28-2009, 10:02 AM

Yes only pbxes from speed dial other i.e callcentric,poivy,voipbuster and other all working fine

martin · 07-28-2009, 10:12 AM

Quote:

Originally Posted by richard99

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?

richard99 · 07-28-2009, 10:30 AM

No. I can not call using dial plan any pbxes extension same ...busy tone. all other provider working normal using dial plan.

emoci · 07-28-2009, 02:03 PM

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...

All4Fun · 07-28-2009, 02:29 PM

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.

richard99 · 07-28-2009, 04:19 PM

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.

emoci · 07-28-2009, 05:07 PM

Quote:

Originally Posted by richard99

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?

richard99 · 07-28-2009, 05:20 PM

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

07-28-2009, 08:39 AM	#1
richard99 Senior Member Join Date: Feb 2007 Location: Indiana, USA Posts: 166 Thanks: 14 Thanked 22 Times in 19 Posts	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.

07-28-2009, 10:30 AM	#5
richard99 Senior Member Join Date: Feb 2007 Location: Indiana, USA Posts: 166 Thanks: 14 Thanked 22 Times in 19 Posts	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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07-28-2009, 10:02 AM	#3
richard99 Senior Member Join Date: Feb 2007 Location: Indiana, USA Posts: 166 Thanks: 14 Thanked 22 Times in 19 Posts	Yes only pbxes from speed dial other i.e callcentric,poivy,voipbuster and other all working fine

07-28-2009, 02:03 PM	#6
emoci Join Date: Jul 2007 Location: Toronto, Canada Posts: 1,422 Thanks: 123 Thanked 369 Times in 282 Posts	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...

07-28-2009, 02:29 PM	#7
All4Fun Junior Member Join Date: Jul 2009 Posts: 22 Thanks: 6 Thanked 3 Times in 2 Posts	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.

07-28-2009, 04:19 PM	#8
richard99 Senior Member Join Date: Feb 2007 Location: Indiana, USA Posts: 166 Thanks: 14 Thanked 22 Times in 19 Posts	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.

07-28-2009, 05:20 PM	#10
richard99 Senior Member Join Date: Feb 2007 Location: Indiana, USA Posts: 166 Thanks: 14 Thanked 22 Times in 19 Posts	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