Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 06:33:35PM -0400, Eric ManxPower Wieling wrote: The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Ah. Got it. Thanks. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
Well, it worked, but ok, I'll take it off.
Now to figure out how to do it across IAX channels from one server to
another.
Which I have, but I haven't tested it yet.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote: On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span Hmmm. Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:14:14PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote: On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span Hmmm. Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+ CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP|
7274514974|2008-07-25 10:14:22
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in
new stack
Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create
channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack
-- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack
-- Executing Hangup(SIP/101cathy-b7619990, ) in new stack
== Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101
cathy-b7619990'
==88
Copied and pasted. I later extended the rules, as you saw, to have a
special rule for 880X, and it worked just fine.
Not sure what to tell you, but it seems to be that.
Note that I have not *yet* taken the -1 off the end, so it cannot be
that.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote: Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 As nearly as I can discern, those are messages where the Zap channel ide is being generated by Asterisk, based on no particular configuration we gave it (there are lots of others, but they could just be repeating an argument they were passed; mostly Application messages). We do in fact, see that zt_request message, but it's not like we made *up* the whole 73-1 thing... :-) Cheers, - jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:28:10PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1'
as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+ CALL LOG START :
|1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP|
7274514974|2008-07-25 10:14:22
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in
new stack
Why do you keep adding that -1?
Try Zap/01
Though I tried originating a call to Zap/04 and Zap/04-1 and both worked
well here (1.4). With the -1 I got the warning I mentioned above about
the unknown option.
Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to
create
channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack
-- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack
-- Executing Hangup(SIP/101cathy-b7619990, ) in new stack
== Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101
cathy-b7619990'
==88
Copied and pasted. I later extended the rules, as you saw, to have a
special rule for 880X, and it worked just fine.
Not sure what to tell you, but it seems to be that.
Note that I have not *yet* taken the -1 off the end, so it cannot be
that.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL
PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:32:34PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote: Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. You can replace the '-1' with 'X56456456', '_123123' or 'p0'. It would be likewise (in)valid, give a warning regarding invalid option but dial anyway. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 3:53 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. You can replace the '-1' with 'X56456456', '_123123' or 'p0'. It would be likewise (in)valid, give a warning regarding invalid option but dial anyway. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir If you want to test inbound and fill all of your channels, you could post something creative on Craigslist and then put them all in a queue with MOH that would keep them on the line. Or you could make a dialplan that takes the inbound caller ID and turn around and dial it. Do that with one of your DIDs and you should fill all your channels pretty quickly. Anyways, with a PRI, when I see the channels come up and I can dial out and in, I have never had an issue with a particular channel. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? Cheers, -- jra -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _88XX1NXXNXX,5,Hangup
exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _880X1NXXNXX,5,Hangup
Which I just retested and it works.
Now to figure out how to do it across IAX channels from one server to
another.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _88XX1NXXNXX,5,Hangup
exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _880X1NXXNXX,5,Hangup
Which I just retested and it works.
Now to figure out how to do it across IAX channels from one server to
another.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL
PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
Quote
seems like a dial-by-span syntax.
What is Dial-by-span ?
I have looked and cannot seem to fund that term.
More likely a comment on my ability to find it than on it obscurity
Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
exten = _88XX1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _88XX1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _88XX1NXXNXX,5,Hangup
exten = _880X1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _880X1NXXNXX,2,Dial(Zap/${EXTEN:3:1}-1/${EXTEN:4},30,o)
exten = _880X1NXXNXX,3,NoOP(${DIALSTATUS})
exten = _880X1NXXNXX,4,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _880X1NXXNXX,5,Hangup
Which I just retested and it works.
Now to figure out how to do it across IAX channels from one server to
another.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL
PROTECTED]
Designer The Things I Think RFC
2100
Ashworth Associates http://baylink.pitas.com '87
e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647
1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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 --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 09:23:44AM -0400, Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I assume I'd have to make a separate group for each channel in the
/etc/asterisk/zapata.conf? Or could I just specify the channel number
directly in the dialplan and make 24 trunkgroups there with a
dialpattern for each one? (I know enough to be dangerous, but not
quite enough to implement without a little help. :-)
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Nice. I assume the Noop's capture the text in the log, then? (See?
Told you I was fresh caught :-)
Hold it: how do I specify the channel? Ah, no, I see what you're
doing. I wanted to actually dial the channel number.
I came up with this:
; dial a long-distance call; allow the user to select a Zap channel
manually
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:3:2}-1/${EXTEN:4},30,o)
exten = _88XX1NXXNXX,3,Hangup
But I'll add the noops.
Course I have to fix the dialplan in my Poly600, too.
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer The Things I Think RFC 2100
Ashworth Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274
Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Josef Stalin)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Acceptance testing of a new PRI
So I have these 4 new PRIs turning up tomorrow. Anyone have any suggestions on some dialplan that I could use to allow me to manually dial calls out over each channel for testing? I assume I'd have to make a separate group for each channel in the /etc/asterisk/zapata.conf? Or could I just specify the channel number directly in the dialplan and make 24 trunkgroups there with a dialpattern for each one? (I know enough to be dangerous, but not quite enough to implement without a little help. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Thursday 24 July 2008 10:30:26 Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten = _71NXXNXX,n,Set(CALLERID(number)=734XXX)
exten = _71NXXNXX,n,Dial(ZAP/${ZAPLINE}/${EXTEN:1})
exten = _71NXXNXX,n,NoOP(${DIALSTATUS})
exten = _71NXXNXX,n,NoOP(Hangup Cause: ${HANGUPCAUSE})
exten = _71NXXNXX,n,Hangup()
Nice. I assume the Noop's capture the text in the log, then? (See?
Told you I was fresh caught :-)
NoOp doesn't capture anything, unless you have Verbose logging turned on and
the verbose level is high enough (3 or higher). If you want direct logging,
use the Log() application in 1.6.
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
