[asterisk-users] PRI (Primary-NTT)
hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code 27. (as usual, phone co. just said it's problem with our equipment without giving us any detail). anybody have any suggestions? here's our /etc/dahdi/system.conf: loadzone=hk defaultzone=hk span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/chan_dahdi.conf: switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=pri_in channel => 1-23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit registration concurrency per friend
Thanks - this is what I needed to know -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Thanks What would you use to measure jitter / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
Sometimes just the act of collecting performance data degrades the quality Sent from my iPhone 5 On Jan 6, 2013, at 6:00 AM, XBrian wrote: > Thanks > > What would you use to measure jitter / packetloss in real time? > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI (Primary-NTT)
Your carrier is apparently using a Japanese switch (at least it's a Japanese standard). If you don't get a good answer from the list, you might send an email to louis.lecl...@denphone.com. Denphone is a company installing Asterisk systems in Tokyo. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edwin Lam Sent: Sunday, January 06, 2013 3:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI (Primary-NTT) hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code 27. (as usual, phone co. just said it's problem with our equipment without giving us any detail). anybody have any suggestions? here's our /etc/dahdi/system.conf: loadzone=hk defaultzone=hk span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/chan_dahdi.conf: switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=pri_in channel => 1-23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Malicious traffic comming from 37.75.210.90
Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Build asterisk for VIA C3
On Thu, Jan 03, 2013 at 10:38:40AM -0500, neo haux wrote: > Is it difficult to publish a build asterisk.deb compiled for VIA > C3 architecture ? Instead of using the binary just for me. > So any one trying to install it on C3 CPU will need just to do: > aptitude install asterisk > > The one that is installed by default doesn't work for such a CPU > > Should I contact debian dev team for that? Any problem with the standard Debian one? Could you please be more specific regarding the versions? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan - working out when users answer
Hi Satish,
Thanks for your response - sorry on the slow reply.
So I've tried the following in the dialplan:
exten =>
direct,n,Dial(${QUEUEEXTS},${RINGTIME},U(queueControl,direct^CONNECTED))
This has a very strange behavior - the NoOp that is in
queueControl,direct,n(CONNECTED) does not show up, however I get the following:
[2013-01-07 17:31:39] ERROR[19135]: app_stack.c:420 gosub_exec: Attempt to
reach a non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)
I've also tried with a macro:
exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME},M(inboundconnected))
[macro-inboundconnected]
exten => s,1,NoOp(Inbound connected!)
It definitely seems like it's being called, but again no NoOp:
-- Executing [direct@queueControl:11] Dial("SIP/1000-47f1",
"SIP/1000,20,M(inboundconnected)") in new stack
I would expect some kind of error if I was doing this wrong - have I missed
something?
Thanks for your or anyone elses help in advance!
Andrew
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 7:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
mailto:and...@computersforall.com.au>> wrote:
Hi Satish/list,
Looks like I spoke to soon.
I have the following in my dialplan:
Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))
And after confirming with a "dialplan show" it was definitely in there, I
continued to get this:
ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a non-existent
destination for gosub: (Context:queueControl, Extension:s, Priority:1)
I can't quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
Any ideas?
Thanks!
In your case, direct and CONNECTED have to be arguments and not the extension
and priority value respectively. Calling Subroutine from dial will always start
execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using ^
as a delimiter.
--Satish Barot
From: Andrew White
Sent: Wednesday, 19 December 2012 5:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Dialplan - working out when users answer
Thanks Satish, fantastic advice. I didn't even think to look into the dial
options - doh!
Thanks very much,
Andrew
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Sent: Wednesday, 19 December 2012 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
mailto:and...@computersforall.com.au>> wrote:
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g.
SIP/100&SIP/101&SIP/105 etc
This works great, however I want to see if I can find a way to work out (and
run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
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