Re: [asterisk-users] Asterisk going down
I'd vote for DNS problem myself. Do you have a local dns sevre that forwards unknow requests? -- Josiah Bryan Productive Concepts, Inc. jbr...@productiveconcepts.com Cell: 765-215-0511 Desk: 765-215-6009 x224 -Original Message- From: Danny Dias Date: Sun, 7 Feb 2010 15:48:19 To: Subject: Re: [asterisk-users] Asterisk going down -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
Paul Chambers wrote: > Josiah Bryan wrote: >> >> Problem is that its crashing for seemingly no reason at all, no errors >> on the console, no logs (that I can find), nothing in /var/lib/messages >> - its puzzeling! Management is screaming like banshees, calls are >> dropping like flies, and all hell is about to break loose if I can't >> stop asterisk from crashing every couple of hours, taking down any >> Zaptel calls with it. >> > That description reminds me of a problem I ran into a while back. One > fan had quietly failed, and the temperature would slowly creep up inside > the box until things started 'acting funny' and the box would lock up > soon after. It'd run fine for 3-4 hours, then just keel over and die. > The logs didn't show anything consistent just before the event. The wierd thing is that its *just* the asterisk process that dies - the rest of the system stays solidly up... > Do you have another PC you can swap the drive and cards into, to try to > rule out hardware instability? could you run lm_sensors? (along with one > of the logging/alarm packages that support it). Well, Paul, it looks like that was indeed the problem (hardware instability.) I came into the office last night after everyone left in order to swap out the RAM in the server - lo and behold, I didn't have any of that type of RAM around (RIMM's ??), so I had to do an emergency hard drive & PCI card transplant to a similar chassis. After a bit of tweaking to get ALSA to work right and the NIC to play nice in the new chassis, asterisk came online and worked beautifully. (And, shockingly enough, the zaptel cards just *worked* - no tweaking needed!) So far, no crashes today (by this time, normally it's crashed two or three times already in a day.) So, we'll see how she runs - If I were a betting man, I'd say that something in that old chassis was going out - probably the RAM as stated before, but not sure. As far as the power supply being "good", I believe it was - didn't check. The server was a re-purposed high-end CAD workstation - the dismal RAM and CPU belie the solid construction of the chassis and the quality of the workmanship in the way the server was put together. Now that I've waxed weird, I'll just say the hardware seems to have been the problem and I'll keep and eye on it. This may have yet saved me from converting over to the callweaver fork - we'll see. :-) Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
Wilton Helm wrote: > One relevant question that hasn't been addressed is whether just the > application is crashing or the whole computer (Linux). > > I would second the hardware idea, with emphasis on generic hardware, > especially RAM. I had a Suse 10 box that kept crashing and doing funny > stuff. I ended up running an extended RAM test on it--one of those > pattern sensitivity tests that takes an hour or two to run. Turned out > that one of the SIMMs I had just bought and installed had a subtle > problem. It would never show up on a straightforward test, but certain > address ranges would fail on one or two of the exotic pattern tests. > > It came from a reputable vendor who does 100% testing themselves, so it > was apparently subtle enough to slip through their test. They took it > back and replaced it. I ran for a few weeks without the module with no > crashes and when I put the replacement in everything was still fine. > > Wilton Just the application crashes. I'll try changing RAM simms to see if that helps. Thanks! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
Doug Lytle wrote: > Josiah Bryan wrote: >> Roderick A. Anderson wrote: >> >> How would I go about pinpointing / diagnosing the hardware fault? Not >> sure exactly what to do with memtest86 - any pointers? >> > A lot of distros have memtest86 as a boot option on the CD/DVD. You > select it and let it run. It'll scan for bad memory. And, shoot lots > of red errors when encountered. If the memory checks fail, you'll know > that you need to replace the chip. Ah I see! Gotcha. I'll try to run that tonite or this weekend then when the plant is closed. Thanks! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
Josiah Bryan wrote:
> David Gibbons wrote:
>>
>> Problem is that its crashing for seemingly no reason at all, no errors
>> on the console, no logs (that I can find), nothing in /var/lib/messages
>> - its puzzeling! Management is screaming like banshees, calls are
>> dropping like flies, and all hell is about to break loose if I can't
>> stop asterisk from crashing every couple of hours, taking down any
>> Zaptel calls with it.
>>
>>
>> I am assuming you have debug turned on so that you can see what's going on
>> when it crashes? If not, open the * console (asterisk -r) and type (core set
>> verbose 100) and (core set debug 100). Then leave the console open so you
>> can see if * was doing anything special when it crashed.
>>
>
> I've ran with verbose quite high lately, but havn't left debug on. Well,
> I just opened console and turned debug on to 100 so we'll wait and see
> what it shows next time it crashes. It's due for another any time now...
>
Alright, latest console output right before latest crash shows:
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'script' logged on from 10.10.9.5
-- Executing [...@playground:1] AGI("Local/9...@playground-604a,2",
"paging-hack.pl") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/paging-hack.pl
> Channel Local/9...@playground-604a,1 was answered.
== Manager 'script' logged off from 10.10.9.5
-- Executing [...@playground:1]
Answer("Local/9...@playground-604a,1", "") in new stack
-- Executing [...@playground:2]
PlayTones("Local/9...@playground-604a,1", "750+440+1030+3000+5000+15000")
in new stack
-- Executing [...@playground:3] Wait("Local/9...@playground-604a,1",
"2") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'script' logged on from 127.0.0.1
-- Executing [...@paging:1] Playback("Local/9...@paging-7883,2",
"beep") in new stack
> Channel Local/9...@paging-7883,1 was answered.
-- Executing [...@playground:1] MeetMe("Local/9...@paging-7883,1",
"951|qaA") in new stack
== Manager 'script' logged off from 127.0.0.1
-- AGI Script paging-hack.pl completed, returning 0
-- Executing [...@playground:2] Goto("Local/9...@playground-604a,2",
"conferences|951|1") in new stack
-- Goto (conferences,951,1)
-- Executing [...@conferences:1]
MeetMe("Local/9...@playground-604a,2", "951|qaA") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '951'
-- Playing 'beep' (language 'en')
[Feb 5 11:29:03] WARNING[24824]: file.c:1204 waitstream_core:
Unexpected control subclass '-1'
-- Executing [...@paging:2] Dial("Local/9...@paging-7883,2",
"Console/dsp") in new stack
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
-- Called dsp
-- ALSA/default answered Local/9...@paging-7883,2
asterisk*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).
[r...@asterisk ~]#
I know that all looks a bit weird, but its related to this problem I had
last September:
http://lists.digium.com/pipermail/asterisk-users/2008-September/217822.html
My extensions.conf has the following notes:
; PAGING HACK
; AGI script: paging-hack.pl is called when user dials 249
; The script puts the user in 951, then calls the Console into
; 951, and starts a fork monitoring the users leg of the call -
; as soon as the user hangs up, the fork automatically
; hangs up the Console.
; ? ? WHY ? ??
; Well, simple, as of version 1.4.21.2 of asterisk,
; when a user dialed 249 and got the Console directly,
; after the user hung up, ringing tone was heard over
; the console until I manually typed 'hangup' in the
; asterisk console - even then, asterisk said 'no calls to hangup'
; The mailing list was no help, so I wrote paaging-hack.pl as a,
; well, a hack to get it to a point where paging still worked.
exten => 951,1,MeetMe(951|qaA)
So, 249 does AGI(paging-hack.pl), and from there, the user and the
Console are dragged into a MeetMe conference for the user to speak
his/her page. (The script doesn't do the hangup on the console actually
- it just leaves the console active for the next page.)
So, anyway, thats the output right before the last crash - any ideas as
to why based on that info?
Thanks!
-josiah
--
Josiah Bryan
IT Manager
Productive Concepts, Inc.
jbr...@productiveconcepts.com
(765) 964-6009, ext. 224
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Re: [asterisk-users] Crash Hard, Crash Often
David Gibbons wrote: > > Problem is that its crashing for seemingly no reason at all, no errors > on the console, no logs (that I can find), nothing in /var/lib/messages > - its puzzeling! Management is screaming like banshees, calls are > dropping like flies, and all hell is about to break loose if I can't > stop asterisk from crashing every couple of hours, taking down any > Zaptel calls with it. > > > I am assuming you have debug turned on so that you can see what's going on > when it crashes? If not, open the * console (asterisk -r) and type (core set > verbose 100) and (core set debug 100). Then leave the console open so you can > see if * was doing anything special when it crashed. > I've ran with verbose quite high lately, but havn't left debug on. Well, I just opened console and turned debug on to 100 so we'll wait and see what it shows next time it crashes. It's due for another any time now... -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
It *is* doing mysql CDR and a whole host of custom AGI scripts. AGI to mudge the CID, AGI to handle receptionist routing/selections, AGI for voicemail (not using builtin vm app) - all the AGI scripts do mysql connections. Would the CDR connection be a problem? -josiah Danny Nicholas wrote: > I've been running 1.4.21.2 on SUSE 11.0 for about 4 months. In my > experience, the fewer database interfaces you can use, the more stable it > will be. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josiah Bryan > Sent: Thursday, February 05, 2009 8:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Crash Hard, Crash Often > > I've been using asterisk for 3+ years now, I love it, but it doesnt love > me back. :-) > > It was crashing frequently and seemingly randomly prior to this latest > upgrade. Not sure what version it was running prior to upgrade (it was > probably an old CVS HEAD from 2+ years go.) Anyway, currently running > 1.4.21.2. > > == Problem == > > Problem is that its crashing for seemingly no reason at all, no errors > on the console, no logs (that I can find), nothing in /var/lib/messages > - its puzzeling! Management is screaming like banshees, calls are > dropping like flies, and all hell is about to break loose if I can't > stop asterisk from crashing every couple of hours, taking down any > Zaptel calls with it. > > I've been thinking of switching over to CallWeaver, but I havn't got > another Zaptel card to plugin to my testing box, so I'd like to just get > Asterisk stabilized right now - but I'm at a loss of even where to start. > > == System Details == > > Running FC3, 2.6.9-1.667 kernel, 32 bit, with 256 MB ram and a 20G hard > drive. I've got two 4-port FXO cards in PCI slots. > > lspci reports: > 02:08.0 Communication controller: Tiger Jet Network Inc. Tiger3XX > Modem/ISDN interface > 02:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX > Modem/ISDN interface > > > [r...@asterisk ~]# cat /proc/cpuinfo > processor : 0 > vendor_id : GenuineIntel > cpu family : 15 > model : 1 > model name : Intel(R) Pentium(R) 4 CPU 1.50GHz > stepping: 2 > cpu MHz : 1483.679 > cache size : 256 KB > fdiv_bug: no > hlt_bug : no > f00f_bug: no > coma_bug: no > fpu : yes > fpu_exception : yes > cpuid level : 2 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm > bogomips: 2924.54 > > > == > > Thanks for any help or advice anyone may have. Cheers! > -josiah > -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash Hard, Crash Often
Roderick A. Anderson wrote: > Doug Lytle wrote: >> Josiah Bryan wrote: >>> I've been using asterisk for 3+ years now, I love it, but it doesnt love >>> me back. :-) >>> >>> >> The first place I usually start is with memtest86 > > Here, here! > > Every time I have had problems with a system (not just Asterisk) > crashing and there is nothing in the logs it turns out to be hardware. > > One slight exception was where a UPS would brownout every so often > causing the system to go out to lunch. Even though there were three > power supplies in that system someone (not me) had _forgot_ to put them > on separate UPS' ... they were all on one. > > Actually, that is hardware, just not in the system case. > > So check your hardware. I must admit, I've suspected hardware as well - the individual FXO "chips" on the two TDM400P have slowly gone dead till I only have four working FXO chips between 8 slots - thats fine, since I only have four POTS lines right now, but still a bit annoying. They are 2 - 3 yrs old, so I guess its just their time. How would I go about pinpointing / diagnosing the hardware fault? Not sure exactly what to do with memtest86 - any pointers? Thanks! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crash Hard, Crash Often
I've been using asterisk for 3+ years now, I love it, but it doesnt love me back. :-) It was crashing frequently and seemingly randomly prior to this latest upgrade. Not sure what version it was running prior to upgrade (it was probably an old CVS HEAD from 2+ years go.) Anyway, currently running 1.4.21.2. == Problem == Problem is that its crashing for seemingly no reason at all, no errors on the console, no logs (that I can find), nothing in /var/lib/messages - its puzzeling! Management is screaming like banshees, calls are dropping like flies, and all hell is about to break loose if I can't stop asterisk from crashing every couple of hours, taking down any Zaptel calls with it. I've been thinking of switching over to CallWeaver, but I havn't got another Zaptel card to plugin to my testing box, so I'd like to just get Asterisk stabilized right now - but I'm at a loss of even where to start. == System Details == Running FC3, 2.6.9-1.667 kernel, 32 bit, with 256 MB ram and a 20G hard drive. I've got two 4-port FXO cards in PCI slots. lspci reports: 02:08.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface [r...@asterisk ~]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 1 model name : Intel(R) Pentium(R) 4 CPU 1.50GHz stepping: 2 cpu MHz : 1483.679 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 2924.54 == Thanks for any help or advice anyone may have. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. jbr...@productiveconcepts.com (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Tim Panton wrote:
> On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
>
>> Tim -
>>
>> Do you have any minimal docs or hints on what hooks the DHTML/JS
>> methods
>> are available for scripting? Something like a quickstart javascript
>> example?
>>
>> I'm great with javascript, but I havn't read thru the Java to figure
>> out
>> the hooks yet - if thats whats needed, I dont mind, but I'd rather
>> hear
>> from the guy who knows best.
>>
>> I'm assuming something like:
>>
>>
>>
>>
>> var applet = [get applet ref];
>>
>> function onDialButtonClick()
>> {
>> var number = myFunctionGetPhoneNumber();
>> applet.connectToServer("my.iax.server.com","user","pass");
>> applet.dial(number);
>> [update UI]
>> }
>>
>> function onHangupClick()
>> { applet.hangupCall();applet.disconnectServer() }
>>
>>
>> Something like that?
>>
>> -josiah
>
>
> It's up to Mexuar to decide if they want to release any pre-existing
> documentation
> (and since it isn't in the .rar I guess they don't intend to at the
> moment).
>
> The easiest thing would be to run JavaDoc over the applet class and
> see what public methods exist.
>
Understood - thanks for your patience with these questions.
Regards,
-josiah
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Re: [asterisk-users] IAX Java Softphone?
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS methods
are available for scripting? Something like a quickstart javascript example?
I'm great with javascript, but I havn't read thru the Java to figure out
the hooks yet - if thats whats needed, I dont mind, but I'd rather hear
from the guy who knows best.
I'm assuming something like:
var applet = [get applet ref];
function onDialButtonClick()
{
var number = myFunctionGetPhoneNumber();
applet.connectToServer("my.iax.server.com","user","pass");
applet.dial(number);
[update UI]
}
function onHangupClick() { applet.hangupCall();applet.disconnectServer() }
Something like that?
-josiah
Tim Panton wrote:
> On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
>
>> Tim Panton ha scritto:
>>> On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
>>>
>>>
>>>> Tim Panton ha scritto:
>>>>
>>>>> It isn't really in a state for novices at the present
>>>>> you'd need:
>>>>> 1) a java compiler
>>>>> 2) a java code signing certificate (java applets can't read from
>>>>> the
>>>>> mic
>>>>> without being signed)
>>>>> 3) appropriate javascript and DHTML to implement the look and feel
>>>>> 4) an asterisk (or freeSWITCH) to talk IAX to.
>>>>>
>>>>> Tim.
>>>>>
>>>>>
>>>> Really great stuff! Could you please explain how to use it in a java
>>>> application?
>>>>
>>>> Thanks in advance.
>>>>
>>> I designed it as a Java applet, so the top level needs Javascript and
>>> DHTML from the
>>> browser to provide a UI.
>>> That said, It wouldn't be very hard to write an application class and
>>> some
>>> UI classes to turn it into a stand-alone application , but that
>>> depends on the
>>> complexity of the UI you want.
>>>
>> I'm interested to use it as IAX2 API within my UI, so something like:
>>
>> - open IAX2 channel
>> - call 123456
>> - answer a call
>> - close IAX2 channel
> It is definitely capable of that with an added class or 2.
> - but remember it is GPL, so you would 'taint' the rest of your code
> - if it isn't already GPL.
>
>
> -
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>
>
>
>
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Re: [asterisk-users] iax2 bindaddress: how to reload so iax2 can bind to an alias IP
Vieri wrote: > --- On Fri, 1/9/09, Tilghman Lesher wrote: >>> --- On Fri, 1/9/09, Josiah Bryan >> wrote: >>>> Is this right after bringing online the alias IP? >>>> If so, you might try using arp-sk to broadcast an >> ARP >>>> packet to kick-start the IP lookup... >>>> http://sid.rstack.org/arp-sk/ >>> Thanks for the link. >>> However, it's not that the two boxes don't >> "see" each other: they can ping, >>> do HTTP, etc. The only service not working is IAX (ie. >> "iax2 show peers" >>> says that the peer is unreachable). So basically, if I >> try to communicate >>> with the alias IP address via several protocols such >> as HTTP, ICMP, SSH, >>> etc, then all works fine. However, if I setup a iax >> trunk, it does not >>> work. >>> I also tested the connection with a softphone (zoiper) >> and it fails too (it >>> fails to connect to the alias IP but it connects fine >> to the nic's >>> "standard" IP address). >>> >>> So, I'm puzzled by the following: >>> >>> These WORK: >>> telnet 80 >>> ssh >>> ping >>> iaxclient >>> >>> But this doesn't: >>> iaxclient >>> >>> even though iax.conf has bindaddr=0.0.0.0 >>> >>> Any clue as to why it's behaving this way? >> Yes. All protocols prior to 1.6.0 use the routing table to >> determine what >> address to use when sending out packets related to a call. >> Normally, the >> alias address is not in the routing table at all, which is >> why it is not >> 'seen' by IAX2. Now, starting in 1.6.0, we have >> the capability within >> Asterisk to use alias addresses within certain protocols, >> but AFAIK, that >> has only been implemented within the SIP stack. > > Thanks for your helpful feedback. > > Curiously, UDP SIP works with alias IP addresses even in 1.2.30. > I guess that the IAX implementation is lagging behind in this particular > field. > With re: to the routing table - it may be a silly question, but is that an arp routing table? If so, could you use arp-sk to force populate the routing table for the alias? -josiah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 bindaddress: how to reload so iax2 can bind to an alias IP
Is this right after bringing online the alias IP? If so, you might try using arp-sk to broadcast an ARP packet to kick-start the IP lookup... http://sid.rstack.org/arp-sk/ -josiah Vieri wrote: > I'm trying to figure out how to reload iax2 (without breaking existing calls) > so it can listen on a new IP address (like "ip addr add local ..."). This > alias IP is added/removed by a custom process (script) for clustering > purposes. > > The iax.conf file contains "bindaddr=0.0.0.0". > > I tried a "iax2 reload" (executed without errors or warnings) but I'm still > not able to connect to the alias IP from another Asterisk server within the > same LAN (no firewalls of any type in the middle). > > On the other Asterisk server I run a "iax2 show peers" and get "status > UNREACHABLE" for the alias IP trunk (eg. 192.168.250.115 in the example > below). > > However, if I specify a non-alias IP address on the other server's trunk > settings then a "iax2 show peers" shows a "status OK" (eg. 192.168.250.111 in > the example below). > > Just for your information, the IP settings on the server I'm trying to > connect to are: > # ip addr show eth1 > 3: eth1: mtu 1500 qdisc pfifo_fast qlen 1000 > link/ether 00:1d:60:b0:25:10 brd ff:ff:ff:ff:ff:ff > inet 192.168.250.111/24 brd 192.168.250.255 scope global eth1 > inet 192.168.250.115/24 brd 192.168.250.255 scope global secondary eth1 > inet6 fe80::21d:60ff:feb0:2510/64 scope link >valid_lft forever preferred_lft forever > > This Asterisk server has version 1.2.30. > > The other "connecting" Asterisk server has version 1.4.21.2. > > Any ideas as to what I could try? > > Vieri > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] For Dial(), when calling party hangs up, redirect called party to another location in the dialplan?
Martin Smith wrote: > Hi all, > > I know when doing a Dial, when the called party hangs up, we have a few > different ways to redirect the calling party to other parts of the > dialplan. > > In this case, I have someone who would like to do the opposite... When > the calling party hangs up after a Dial(), redirect the called party to > another location. > > I'm not sure how else to describe what the user wants to do, but I'm > willing to try if people have questions :) > > Is there a simple way to do this without a meetme room? Umm,, I can't think of any way other than a Meetme room and a healthy sprinkling of AGI... -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ldap usage in 1.6.0
Brendan Martens wrote: > > Having thought some more about my issue I think I can perhaps ask my > question more succinctly: is it possible to get dynamic (or > "realtime") data from ldap within the various .conf files? > > If there is not a convenient function for getting this in the .conf > files, what if I somehow specified a global variable within the > res_ldap.conf and referenced that value inside the other .conf files? > Is this possible? Sorry if these are very basic questions, I just > haven't been able to find answers to them. : ( > Here, I've written a perl script that rewrites the actual sip.conf itself (as well as generates a custom myexten.conf file, which is included in the main extensions.conf file.) The perl script can read from whatever datasource you setup - right now, I read from a MySQL database of users, but I know perl can read from an LDAP directory as well. This way, perl sits between Asterisk and the database/directory and does the mapping/translation required, giving more complete control over the sync process. Of course, I wrote this script in the pre-1.2 days of Asterisk, but its still running fine on 1.4.* that I've got in production right now. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Missing 'Queue' Application in 1.4.21.2
Gotta love flukes - after stopping asterisk and restarting so I could see the startup text, "core show application Queue" just worked . ??? Oh well. Thanks! -josiah Andres wrote: > Josiah Bryan wrote: > >> Hey All - >> >> Slight problem here - my install of 1.4.21.2 seems to be "missing" the >> Queue application: >> >> > What does the CLI output say when you start asterisk and it gets to the > part where it tries to load app_queue.so? > > Andres > http://www.neuroredes.com > >> asterisk*CLI> core show version >> Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a >> i686 running Linux on 2008-09-02 18:15:03 UTC >> >> asterisk*CLI> core show application Queue >> Your application(s) is (are) not registered >> >> I checked 'make meuselect' and it *seems* to indicate that app_queue was >> built: >> >> >>** >>Asterisk Module and Build Option Selection >>** >> >>Press 'h' for help. >> >> [*] 37. app_nbscat >> XXX 38. app_osplookup >> [*] 39. app_page >> [*] 40. app_parkandannounce >> [*] 41. app_playback >> [*] 42. app_privacy >> [*] 43. app_queue >> [*] 44. app_random >> [*] 45. app_read >> [*] 46. app_readfile >> ... More ... >> >> >> True Call Queueing >> Depends on: res_monitor(M) >> >> >> Any ideas on how to figure this out? >> >> Many thanks, >> -josiah >> >> >> > > > ___ > -- 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 > -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Missing 'Queue' Application in 1.4.21.2
Hey All - Slight problem here - my install of 1.4.21.2 seems to be "missing" the Queue application: asterisk*CLI> core show version Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a i686 running Linux on 2008-09-02 18:15:03 UTC asterisk*CLI> core show application Queue Your application(s) is (are) not registered I checked 'make meuselect' and it *seems* to indicate that app_queue was built: ** Asterisk Module and Build Option Selection ** Press 'h' for help. [*] 37. app_nbscat XXX 38. app_osplookup [*] 39. app_page [*] 40. app_parkandannounce [*] 41. app_playback [*] 42. app_privacy [*] 43. app_queue [*] 44. app_random [*] 45. app_read [*] 46. app_readfile ... More ... True Call Queueing Depends on: res_monitor(M) Any ideas on how to figure this out? Many thanks, -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] How to add Callee's name into Dial command ?
Mr Shunz wrote:
> [snip]
>
>> To make myself clear, here is what I'm trying to do : when Alice is calling
>> Bob (Alice ---> Asterisk--->Bob), I would like Bob's phone to
>> display Alice's name (no problem, for that) but I would also like Alice's
>> phone screen to display Bob's name (instead of Bob's number)
>
> mmm ... this wasn't clear on your OP ...
> so you need to show the CALLED name on the CALLER phone ...
>
>> My SIP hardphone is capable of displaying P-Asserted-Identity in outbound
>> calls (not just inbound) but I
>> couldn't find any way to teach Asterisk to fill this P-Asserted-Identity
>> header :
>
> you can try with: (*** untested ***)
>
> exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:
> '${CALLERID(name)' <${CALLERID(num)>)
Interesting idea - I can see that being very useful. I know on my old
SPPA-841s they would do that - but it was based on looking up the dialed
number in the internal directory (which I programmed using a perl script
in the asterisk server.) So, when I dialed 213, the name appeared that I
dialed, confirming I had the right person.
Unfortunately, I never have been able to get my Polycom SoundPoint IP
500 phones to do that.
I just tested the SIPAddHeader command given above - doesn't work with
the Polycom Soundpoint IP 500 that I tested with. (Even with the missing
'}' at the end that I fixed - still doesn't work.)
Good idea thought - anybody have any magic that might make that work?
-josiah
--
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [asterisk-users] Asterisk Documentation now on voip-info.org Wiki
Rob Hillis wrote: > Josiah Bryan wrote: >> Any formatting can be added as desired - this was just a quick way to >> get the content online. >> > > Might I suggest including... > > print "-=NOTE: These pages are automatically updated once per > day/week/month/year/decade from the Asterisk subversion repository. Any > changes made to this page will be automatically overwritten with the > latest version from .\n"; > > ...at the beginning? May stop some nutters whining that you're > continually overwriting their changes. > Good point - I'll get that in there after breakfast :-). Seriously, good point though. -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk Documentation now on voip-info.org Wiki
Tzafrir Cohen wrote: > On Tue, Sep 30, 2008 at 01:35:00PM -0400, Josiah Bryan wrote: >> Tzafrir Cohen wrote: >>> On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote: >>>> Hey All - >>>> >>>> Link to the index page: >>>> >>>> http://www.voip-info.org/wiki/view/Asterisk+Documentation >>>> >>> Why not link to the SVN instead? >> I considered that as well. My thoughts: >> >> 1) Ungoogleabelness (if thats a word :-) - since google already ranks >> voip-info.org high on search for asterisk related content, I thought the >> docs should be where the users are, not vis-a-versa. >> >> 2) Formatability - the docs are plain text in subversion, whereas >> putting the in the wiki offers the possibility for formatting and >> auto-linking as the algorithm presents itself. > > Do you intend to add that formatting in your script? They can't be > changed manually. The script design supports plugin formatting as it stands. E.g. I can insert any formatting algorithm if anyone has any suggestions. Right now, the formatter script just does: #!/usr/bin/perl use strict; my $file = $ARGV[0]; print "~pp~\n"; print `cat $file`; print "~/pp~\n"; Any formatting can be added as desired - this was just a quick way to get the content online. >> 2) UI similarity - linking to the file on svn, for example: >> >> http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co >> >> Brings just the plain text view, whereas putting it in the wiki offers >> the same UI as the rest of the site. > > http://svn.digium.com/svn/asterisk/branches/1.6.0/doc/callfiles.txt > > Looks better. I agree - if you're looking for the change log. However, I (if I were a first-time asterisk user) probably don't care for the change-log-esque view, I just want to read the text for myself. However, I'd be happy to add links to the svn at the bottom of the page if that is desired. Thoughts? Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk Documentation now on voip-info.org Wiki
Tzafrir Cohen wrote: > On Tue, Sep 30, 2008 at 11:32:41AM -0400, Josiah Bryan wrote: >> Hey All - >> >> Link to the index page: >> >> http://www.voip-info.org/wiki/view/Asterisk+Documentation >> > Why not link to the SVN instead? I considered that as well. My thoughts: 1) Ungoogleabelness (if thats a word :-) - since google already ranks voip-info.org high on search for asterisk related content, I thought the docs should be where the users are, not vis-a-versa. 2) Formatability - the docs are plain text in subversion, whereas putting the in the wiki offers the possibility for formatting and auto-linking as the algorithm presents itself. 2) UI similarity - linking to the file on svn, for example: http://svn.digium.com/view/asterisk/branches/1.6.0/doc/callfiles.txt?view=co Brings just the plain text view, whereas putting it in the wiki offers the same UI as the rest of the site. Note that all these comments are merely my thoughts - feel free to comment against them at will. If desired, I can update the index page generator to just put links to the svn instead. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk Documentation now on voip-info.org Wiki
Hey All - Per a discussion earlier, I've setup a small cron job on one of my servers that automatically updates voip-info.org wiki with the latest and greatest Asterisk Documentation, straight from svn (specifically, the /branches/$version/doc folder for each version.) The files are located under 'Asterisk Documentation' on voip-info.org: Link to the index page: http://www.voip-info.org/wiki/view/Asterisk+Documentation It currently polls the following Asterisk branches from subversion: * 1.2 * 1.4 * 1.6.0 * 1.6.1 --- Let me know what you think. If anyone has any questions or comments, please do let me know. Oh, and many thanks to James Thompson of voip-info.org for his quick response to my questions about an API for updating pages. His help was invaluable. --- Technical Specifics about the Cron Job: --- The cron job runs daily (about 4am EST) and does an 'svn update' for each version's 'doc' folder. If there are any changes, the job uploads ONLY the 'text/plain' files in the folder to the wiki (prefixing the pages with 'Asterisk Documentation '+$version+' '+$filename, so /branches/1.6.1/doc/callfiles.txt becomes 'Asterisk Documentation 1.6.1 callfiles.txt': http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt Note that right now, the files are just passed straight to the wiki and quoted in a '~pp~' block (essentially, a block) - formatting can be applied later if requested - and if presented with a reliable formatting algorithm. Let me know what you all think. Cheers! -josiah -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk Documentation on voip-info.org (was: Re: Knowing incoming call technology and channel [SOLVED])
I've started the page at: http://www.voip-info.org/wiki-Asterisk+Documentation But I'm having problems with logging in via a script - I emailed [EMAIL PROTECTED] and J. Thompson has been very responsive in my request for help. I'll post back here when I've got something online to show. Cheers! -josiah Eric "ManxPower" Wieling wrote: > You would want three pages, 1.2 docs, 1.4 docs, and 1.6 docs. > > Mark Hamilton wrote: >> I don't see why not, Voip-info is very outdated in most respects. >> Most of it with bad examples, dating to Asterisk 1.x era. >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan >> Sent: September 29, 2008 11:09 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Knowing incoming call technology and channel >> [SOLVED] >> >> So, should we (I can do it, if desired) write a script that polls >> subversion docs directory and imports it into voip-info.org when the the >> docs are changed? >> >> I'd be glad to write and host such a script if the community desires the >> feature. >> >> -josiah >> >> SIP wrote: >>> Eric "ManxPower" Wieling wrote: >>>> Olivier wrote: >>>> >>>> >>>>> I don't have any spare zaptel enabled system I could try this on, but I >>>>> was not aware of this CHANNEL variable. >>>>> Now, I can see it here >> http://www.voip-info.org/wiki/view/Asterisk+variables >>>>> Maybe, I will add a line in www.voip-info.org <http://www.voip-info.org> >>>>> to keep others (me?) from searching again. >>>>> >>>> You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt >>>> There's lots of cool information there, and all of it is up to date >>>> for your version of Asterisk, unlike voip-info.org. >>>> >>>> I often wonder why nobody seems to read the docs that are included with >>>> Asterisk. >>>> >>>> >>> Web and/or context-searchable documentation will ALWAYS win out over a >>> somewhat loose collection of text files. >>> >>> That's basic UI psychology 101. >>> >>> N. >>> >>> ___ >>> -- 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 >>> > -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Zaptel Lines - How many are in use..
I normally use 'core show channels' and check for 'Zap/' in the channel string. Are you trying to do it in an automated way, like from AGI? Singer Wang wrote: > Hello, > > I have a question. We have a 8 port FXO card in our asterisk server > plugged into 8 analog lines. Is there a way to tell at how many of those > ports are in use (AKA, actually on a call)? I tried zap show status and > zap show channel [num] but I don't see anything that might be helpful. > > Singer > > -- > *Singer X.J. Wang* > /Systems Engineer/ > The Pythian Group > > Office: (613) 565-8696 x298 > Toll Free:(877) 798-4426 x298 > Cell: (613) 218-9184 > Fax: (613) 565-8710 > Email:[EMAIL PROTECTED] > MSN: [EMAIL PROTECTED] > Yahoo:pythianwang > AIM: pythianwang > ICQ: 201253 > Gadu-Gadu:6817795 > Tencent QQ: 858310404 > > ___ > -- 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 -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Knowing incoming call technology and channel [SOLVED]
So, should we (I can do it, if desired) write a script that polls subversion docs directory and imports it into voip-info.org when the the docs are changed? I'd be glad to write and host such a script if the community desires the feature. -josiah SIP wrote: > Eric "ManxPower" Wieling wrote: >> Olivier wrote: >> >> >>> I don't have any spare zaptel enabled system I could try this on, but I >>> was not aware of this CHANNEL variable. >>> Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables >>> Maybe, I will add a line in www.voip-info.org <http://www.voip-info.org> >>> to keep others (me?) from searching again. >>> >> You should have looked in /path/to/arc/asterisk/doc/channelvariables.txt >> There's lots of cool information there, and all of it is up to date >> for your version of Asterisk, unlike voip-info.org. >> >> I often wonder why nobody seems to read the docs that are included with >> Asterisk. >> >> > Web and/or context-searchable documentation will ALWAYS win out over a > somewhat loose collection of text files. > > That's basic UI psychology 101. > > N. > > ___ > -- 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 > -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk - Operator switch billing
A simple AGI script would be able to handle that easily, I would think. Or am I missing something in the details? -josiah Sriram wrote: > Hi All > > I am a premium IVR content service provider thats runs on premium rate > lines, my setup (currently on PRIs) is like customer dials the short > code (premium number) which gets forwarded on the PRIs to my IVR. In > the normal world the customer is charged immediately the call is > answered by the IVR. On operator's new requirement - he wants me to > design the IVR in such a way that a customer will call on a number (Toll > Free/Premium Rate) but the billing will start only if his MSISDN is > present on a database that he will give it to me...Is this sort of > differential charging on a single call possible in Asterisk ? If yes how > and what additional parameters do i need to get from him > > Please assist > > Thanks > Sriram > > > > > > > ___ > -- 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] Ringing on Console after a page
Good questions - the only answer to the Goto is that this was a legacy dialplan that I first wrote 3+ years ago when I first set up asterisk - and I havn't gone go back and re-work it after learning more about asterisk - it worked up till the upgrade to 1.4 and that was that. However, you're right - simpler is better anyway. I changed it to the 249,n,Dial(Console/dsp) format (as you described below) and it still plays the ringing indicator over the console after I hangup my phone. As an aside, In the 3+ years that the system has been online, users know that when they dialed 249 and heard "Goodbye!" right away, they weren't going to be able to page and "Something was wrong." (Usually, someone had put 249 on hold or something like that.) Thats the primary reason I left the goodbye in there. Anyway, thoughts on how to debug? Thanks for your help and your suggestions. -josiah Barry L. Kline wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Josiah Bryan wrote: > >> [paging] >> exten => 249,1,Goto(paging,s,1) >> exten => s,1,Playback(beep) >> exten => s,n,Dial(Console/dsp) >> exten => s,n,Playback(vm-goodbye) >> exten => s,n,Hangup > > If the caller has hung up, to whom are you playing the "vm-goodbye" > message? Also, why the Goto? > > [paging] > exten => 249,1,Playback(beep) > exten => 249,n,Dial(Console/dsp) > > Barry > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.5 (GNU/Linux) > > iD8DBQFIvvVQCFu3bIiwtTARAp9XAJ0Ra8LLo2COS89loyBFgWutV5SxcgCbB6Md > 54ve7snza6SLYZ1ufR4BVJY= > =Y8MF > -END PGP SIGNATURE- > > ___ > -- 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 > -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] Asterisk Crash
Well, at the time I wrote the AGI, fewestcalls wasn't an option (or at least, I couldn't find it through googling or on the voip-info wiki). Since then, the script has been in production use for 3+ years and I havn't bothered to go back rework the dialplan. Sorry for the trouble though. However, it still begs the question, why does Dial seem to "fall through" like that after the operator transfers the call? Is that expected/designed behavior? If yes, Has that changed since the 1.0 days of asterisk? If yes, Is there a switch that can turn that off? Thanks for your patience with all these questions. Regards, -josiah Andrew Latham wrote: > I know many are thinking this but why don't you use a queue with > "fewestcalls" for the strategy? > > > On Wed, Sep 3, 2008 at 4:04 PM, Josiah Bryan > <[EMAIL PROTECTED]> wrote: >> Alright, praise , I think I've got an idea on *what* its crashing >> on- I've tested the change below and Asterisk no longer crashes at that >> point. I'm crosing my fingers hoping that it doesn't crash anywhere else. >> >> Bottom line: Asterisk is crashing when AGI tells it to 'Dial' for an >> active call that the operator just transfered. >> >> Details: I've got an AGI script that routes the call to one of three >> receptionists based on call load for that SIP device (uses manager to >> show channels to get a count of how many calls each operator is handling >> at that moment. Got that so far? Okay. >> >> Here's the basic flow: The AGI script figures out which SIP device to >> send the call to. It sends an agi exec command to Asterisk to Dial >> $device|45. Operator answers call, does her script. Operator then >> presses transfer button on her phone to transfer to whomever the call is >> destined for. Somewhere after the time she presses transfer, asterisk >> seems to go back to executing the AGI script. >> >> Now, the next command after the Dial $device|45 is another 'Dial' to a >> backup operator (normally myself or another person in the IT department) >> - that way we can be sure the call gets to a person even if the >> operators don't answer. Anyway, as soon as the second Dial is executed >> to the backup SIP device (whoever is on duty at that time), asterisk >> crashes and burns. >> >> What it seems is that even though the call went thru AGI, AGI dialed the >> operator, operator xfered to dest exten - as soon as the operator let go >> of the call (by xfering), asterisk let the AGI continue...almost as if >> the operator didn't answer. >> >> Normally, in my pre-1.4 environment, the second Dial in the AGI was >> rarely reached - it was only reached if the first operator didn't answer >> - or if the call was hungup, before SIGHUP was received. >> >> But now, post-1.4, it seems to "fall through" so to speak as soon as the >> operator transfers the call. >> >> As a work around, I've just disabled the second Dial command. All the >> calls on my server just cleared out and I just tested it - and yes, that >> was indeed the problem. >> >> Any ideas why Dial seems to fall thru after the operator xfers the call >> or if I can even do anything about that? >> >> Thanks for your help and your time with all of this. >> >> Cheers! >> -josiah >> >> >> Josiah Bryan wrote: >>> "From" was a CVS-HEAD version from way back pre 1.2 days, sometime in >>> the 1.0 days (I think.) >>> >>> I've reviewed my dialplan based on the and UPGRADE.txt notes (and >>> UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, >>> etc.) - really not much was affected in the dialplan. I'm just doing the >>> basic "calls come in to receptionist, she transfers to users extensions" >>> paradigm. >>> >>> yum update is still doing header downloads for the upgrade transaction, >>> and I havn't seen a kernel update come through yet - I'll keep an eye out. >>> >>> As far as DAHDI - didn't know that - googling turned up the digium blog >>> on the topic, but the linked page >>> (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did >>> use a fresh build of zaptel-1.4 (svn r4506) from >>> http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. >>> >>> My watchdog process still is reporting frequent crashes of asterisk >>> (most recent at 12:35 EST - they are on average an hour or less apart - &
[asterisk-users] Ringing on Console after a page
Hello, all -
Alright, after my fun with Asterisk crashing, I'm onto my next item in
my checklist of stuff-to-fix-after-upgrading. I've noticed a very
troubling problem when "paging" over Console/dsp.
(I'm not sure if this has anything to do with the "Dial" oddities that I
experienced with the "Crashing" problem in my other thread or not...)
The problem is that after the user dials the extension, connects, speaks
their page, hangsup, ringing is heard over the paging system (as in, the
tone heard when you dial a person and you hear the phone ringing - that
ringing tone - I don't know the "proper" term for it, but you get the
drift.)
I've gone through the source code, trying to figure out what it could be
doing - however, since this is the first time I've really looked at the
source for asterisk, I really didn't know what to look for.
Here's the relevant context (which is included in a general context for
all users):
[paging]
exten => 249,1,Goto(paging,s,1)
exten => s,1,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Hangup
Here's the console output when I dial extension 249 to page. (I dial,
paging answers, I say whatever (or even just hangup immediately) - then,
right after the call termination, I hear the ringing over the paging
system. I have to *manually* issue then hangup command seen below to
stop it from ringing - however, the oddest thing is asterisk tells me
that there is no call to hangup. Its not like the console got transfered
to any extension - literally no channels active while the ringing is
taking place (core show channels reports 0 active channels even while
the ringing is heard.)
asterisk*CLI> set verbose 99
Verbosity is at least 99
-- Zap/1-1 answered SIP/236-09f0ea20
asterisk*CLI> set debug 99
Core debug was and is now 99
asterisk*CLI>
-- Executing [EMAIL PROTECTED]:1] Goto("SIP/josiah2-09f0ea20",
"paging|s|1") in new stack
-- Goto (paging,s,1)
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/josiah2-09f0ea20", "beep")
in new stack
-- Playing 'beep' (language 'en')
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/josiah2-09f0ea20",
"Console/dsp") in new stack
<< Call placed to 'dsp' on console >>
<< Auto-answered >>
-- Called dsp
-- ALSA/default answered SIP/josiah2-09f0ea20
<< Hangup on console >>
== Spawn extension (paging, s, 2) exited non-zero on
'SIP/josiah2-09f0ea20'
Really destroying SIP dialog '[EMAIL PROTECTED]'
Method: BYE
asterisk*CLI> hangup
No call to hangup up
I'm open to any and all suggestions.
Thanks for your time and patience!
--
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [asterisk-users] Asterisk Crash
Alright, praise , I think I've got an idea on *what* its crashing on- I've tested the change below and Asterisk no longer crashes at that point. I'm crosing my fingers hoping that it doesn't crash anywhere else. Bottom line: Asterisk is crashing when AGI tells it to 'Dial' for an active call that the operator just transfered. Details: I've got an AGI script that routes the call to one of three receptionists based on call load for that SIP device (uses manager to show channels to get a count of how many calls each operator is handling at that moment. Got that so far? Okay. Here's the basic flow: The AGI script figures out which SIP device to send the call to. It sends an agi exec command to Asterisk to Dial $device|45. Operator answers call, does her script. Operator then presses transfer button on her phone to transfer to whomever the call is destined for. Somewhere after the time she presses transfer, asterisk seems to go back to executing the AGI script. Now, the next command after the Dial $device|45 is another 'Dial' to a backup operator (normally myself or another person in the IT department) - that way we can be sure the call gets to a person even if the operators don't answer. Anyway, as soon as the second Dial is executed to the backup SIP device (whoever is on duty at that time), asterisk crashes and burns. What it seems is that even though the call went thru AGI, AGI dialed the operator, operator xfered to dest exten - as soon as the operator let go of the call (by xfering), asterisk let the AGI continue...almost as if the operator didn't answer. Normally, in my pre-1.4 environment, the second Dial in the AGI was rarely reached - it was only reached if the first operator didn't answer - or if the call was hungup, before SIGHUP was received. But now, post-1.4, it seems to "fall through" so to speak as soon as the operator transfers the call. As a work around, I've just disabled the second Dial command. All the calls on my server just cleared out and I just tested it - and yes, that was indeed the problem. Any ideas why Dial seems to fall thru after the operator xfers the call or if I can even do anything about that? Thanks for your help and your time with all of this. Cheers! -josiah Josiah Bryan wrote: > "From" was a CVS-HEAD version from way back pre 1.2 days, sometime in > the 1.0 days (I think.) > > I've reviewed my dialplan based on the and UPGRADE.txt notes (and > UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, > etc.) - really not much was affected in the dialplan. I'm just doing the > basic "calls come in to receptionist, she transfers to users extensions" > paradigm. > > yum update is still doing header downloads for the upgrade transaction, > and I havn't seen a kernel update come through yet - I'll keep an eye out. > > As far as DAHDI - didn't know that - googling turned up the digium blog > on the topic, but the linked page > (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did > use a fresh build of zaptel-1.4 (svn r4506) from > http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. > > My watchdog process still is reporting frequent crashes of asterisk > (most recent at 12:35 EST - they are on average an hour or less apart - > some 5 or 10 minutes apart.) > > Suggestions for further debugging? /var/log/asterisk shows a bunch of > log files - event_log is blank, messages is just warnings from the > console - but NOTHING in /var/log/asterisk/messages from around the > crash times (e.g. at 12:35 EST in messages there is nothing, last msg > was at 12:01 and next msg was at 12:36 indicating a restart of asterisk > with "cdr.c: CDR simple logging enabled" message.). > > Any way to get asterisk to tell me *why* or what app is causing the > crash or termination? > > Thanks for your help with this mess. Cheers! > -josiah > > > > Andrew Latham wrote: >> I have had issues with 2.6.9 in the past but it sounds like that is >> not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. >> If you upgraded from 1.2 did you check your dialplan to see if the >> commands are depreciated and you also understand that a lot has change >> on zaptel which is now DAHDI >> >> >> >> >> On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan >> <[EMAIL PROTECTED]> wrote: >>> Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 >>> (kernel 2.6.9-1.667). (System output of uname -a and more is below the >>> closing.) >>> >>> I've got two wctdm PCI cards running 4 FXO modules
Re: [asterisk-users] Asterisk Crash
"From" was a CVS-HEAD version from way back pre 1.2 days, sometime in the 1.0 days (I think.) I've reviewed my dialplan based on the and UPGRADE.txt notes (and UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, etc.) - really not much was affected in the dialplan. I'm just doing the basic "calls come in to receptionist, she transfers to users extensions" paradigm. yum update is still doing header downloads for the upgrade transaction, and I havn't seen a kernel update come through yet - I'll keep an eye out. As far as DAHDI - didn't know that - googling turned up the digium blog on the topic, but the linked page (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did use a fresh build of zaptel-1.4 (svn r4506) from http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. My watchdog process still is reporting frequent crashes of asterisk (most recent at 12:35 EST - they are on average an hour or less apart - some 5 or 10 minutes apart.) Suggestions for further debugging? /var/log/asterisk shows a bunch of log files - event_log is blank, messages is just warnings from the console - but NOTHING in /var/log/asterisk/messages from around the crash times (e.g. at 12:35 EST in messages there is nothing, last msg was at 12:01 and next msg was at 12:36 indicating a restart of asterisk with "cdr.c: CDR simple logging enabled" message.). Any way to get asterisk to tell me *why* or what app is causing the crash or termination? Thanks for your help with this mess. Cheers! -josiah Andrew Latham wrote: > I have had issues with 2.6.9 in the past but it sounds like that is > not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. > If you upgraded from 1.2 did you check your dialplan to see if the > commands are depreciated and you also understand that a lot has change > on zaptel which is now DAHDI > > > > > On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan > <[EMAIL PROTECTED]> wrote: >> Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 >> (kernel 2.6.9-1.667). (System output of uname -a and more is below the >> closing.) >> >> I've got two wctdm PCI cards running 4 FXO modules each: >> pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F >> pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F >> >> As far as FC3 - I believe last yum update was ran on 6/01 of this year - >> - good suggestion, I'll re-run it right now as I type this...okay, yum >> update running. >> >> The only dmesg output that even looks odd is: >> post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747) >> post_create: setxattr failed, rc=28 (dev=dm-0 ino=451748) >> >> Other than that, the only other dmesg output since reboot (this morning >> 8am or so) is some selinux deny messages related to snmpd and httpd. >> >> >> Suggestions? Thank you for taking the time to look at all of this. >> >> Regards, >> -josiah >> >> >> Here's uname, free, and /proc/cpuinfo: >> >> [EMAIL PROTECTED] asterisk]# uname -a >> Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25 >> EST 2004 i686 i686 i386 GNU/Linux >> >> [EMAIL PROTECTED] asterisk]# free >> total used free sharedbuffers cached >> Mem:255652 253416 2236 0 1380 81220 >> -/+ buffers/cache: 170816 84836 >> Swap: 524280 8340 515940 >> >> [EMAIL PROTECTED] asterisk]# cat /proc/cpuinfo >> processor : 0 >> vendor_id : GenuineIntel >> cpu family : 15 >> model : 1 >> model name : Intel(R) Pentium(R) 4 CPU 1.50GHz >> stepping: 2 >> cpu MHz : 1483.674 >> cache size : 256 KB >> fdiv_bug: no >> hlt_bug : no >> f00f_bug: no >> coma_bug: no >> fpu : yes >> fpu_exception : yes >> cpuid level : 2 >> wp : yes >> flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca >> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm >> bogomips: 2924.54 >> >> >> >> Andrew Latham wrote: >>> What type of hardware are you using? When is the last time you upgraded >>> Fedora? >>> >>> "core set verbose 6" should get you anything you need. Have a look at >>> the "dmesg" output. >>> >>> >>> >>> On Wed, Sep 3, 2008 at 10:14 AM, Josiah Br
Re: [asterisk-users] Asterisk Crash
Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3
(kernel 2.6.9-1.667). (System output of uname -a and more is below the
closing.)
I've got two wctdm PCI cards running 4 FXO modules each:
pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F
pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F
As far as FC3 - I believe last yum update was ran on 6/01 of this year -
- good suggestion, I'll re-run it right now as I type this...okay, yum
update running.
The only dmesg output that even looks odd is:
post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747)
post_create: setxattr failed, rc=28 (dev=dm-0 ino=451748)
Other than that, the only other dmesg output since reboot (this morning
8am or so) is some selinux deny messages related to snmpd and httpd.
Suggestions? Thank you for taking the time to look at all of this.
Regards,
-josiah
Here's uname, free, and /proc/cpuinfo:
[EMAIL PROTECTED] asterisk]# uname -a
Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25
EST 2004 i686 i686 i386 GNU/Linux
[EMAIL PROTECTED] asterisk]# free
total used free sharedbuffers cached
Mem:255652 253416 2236 0 1380 81220
-/+ buffers/cache: 170816 84836
Swap: 524280 8340 515940
[EMAIL PROTECTED] asterisk]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 1
model name : Intel(R) Pentium(R) 4 CPU 1.50GHz
stepping: 2
cpu MHz : 1483.674
cache size : 256 KB
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception : yes
cpuid level : 2
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm
bogomips: 2924.54
Andrew Latham wrote:
> What type of hardware are you using? When is the last time you upgraded
> Fedora?
>
> "core set verbose 6" should get you anything you need. Have a look at
> the "dmesg" output.
>
>
>
> On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan
> <[EMAIL PROTECTED]> wrote:
>> Hello, folks -
>>
>> Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the
>> 'asterisk' process. I thought it was due to mpg123 and music on hold -
>> so I disabled all MOH classes in musiconhold.conf - but still random
>> crashing!
>>
>> Here's a transcript from the console. Right at the "Disconnected"
>> message, the asterisk process had crashed. I've got a watchdog that
>> automatically restarts the process, but that still means all calls were
>> lost.
>>
>> Any advice on how to troubleshoot or diagnose??
>>
>> Thanks!
>> -josiah
>>
>>
>>
>> asterisk*CLI> set verbose 99
>> Verbosity was 1 and is now 99
>> The 'set verbose' command is deprecated and will be removed in a future
>> release. Please use 'core set verbose' instead.
>> -- Music class default requested but no musiconhold loaded.
>> -- Executing [EMAIL PROTECTED]:1] Macro("SIP/op-1-0902f218",
>> "stdexten|213|SIP/213") in new stack
>> -- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/op-1-0902f218",
>> "1?999|1") in new stack
>> -- Goto (macro-stdexten,999,1)
>> -- Executing [EMAIL PROTECTED]:1] Set("SIP/op-1-0902f218",
>> "opt=m") in new stack
>> -- Executing [EMAIL PROTECTED]:2] BackGround("SIP/op-1-0902f218",
>> "transfer") in new stack
>> -- Playing 'transfer' (language 'en')
>> -- Executing [EMAIL PROTECTED]:3] Goto("SIP/op-1-0902f218",
>> "s|dial") in new stack
>> -- Goto (macro-stdexten,s,3)
>> -- Executing [EMAIL PROTECTED]:3] Dial("SIP/op-1-0902f218",
>> "SIP/213|20|m") in new stack
>> -- Called 213
>> -- Music class default requested but no musiconhold loaded.
>> -- AGI Script Executing Application: (Dial) Options: (SIP/201|30)
>> -- SIP/213-090126f8 is ringing
>> asterisk*CLI>
>> Disconnected from Asterisk server
>> Executing last minute cleanups
>> Asterisk cleanly ending (0).
>>
>> --
>> Josiah Bryan
>> IT Manager
>> Productive Concepts, Inc.
>> [EMAIL PROTECTED]
>> (765) 964-6009, ext. 224
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon
[asterisk-users] Asterisk Crash
Hello, folks -
Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the
'asterisk' process. I thought it was due to mpg123 and music on hold -
so I disabled all MOH classes in musiconhold.conf - but still random
crashing!
Here's a transcript from the console. Right at the "Disconnected"
message, the asterisk process had crashed. I've got a watchdog that
automatically restarts the process, but that still means all calls were
lost.
Any advice on how to troubleshoot or diagnose??
Thanks!
-josiah
asterisk*CLI> set verbose 99
Verbosity was 1 and is now 99
The 'set verbose' command is deprecated and will be removed in a future
release. Please use 'core set verbose' instead.
-- Music class default requested but no musiconhold loaded.
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/op-1-0902f218",
"stdexten|213|SIP/213") in new stack
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/op-1-0902f218",
"1?999|1") in new stack
-- Goto (macro-stdexten,999,1)
-- Executing [EMAIL PROTECTED]:1] Set("SIP/op-1-0902f218",
"opt=m") in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround("SIP/op-1-0902f218",
"transfer") in new stack
-- Playing 'transfer' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto("SIP/op-1-0902f218",
"s|dial") in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [EMAIL PROTECTED]:3] Dial("SIP/op-1-0902f218",
"SIP/213|20|m") in new stack
-- Called 213
-- Music class default requested but no musiconhold loaded.
-- AGI Script Executing Application: (Dial) Options: (SIP/201|30)
-- SIP/213-090126f8 is ringing
asterisk*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).
--
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] connecting a sipura sip device to asterisk before dialing any digits
On Wednesday 18 May 2005 12:46 pm, Jon Gabrielson wrote: > I would like the ability for a sipura sip device to > instantly connect to an asterisk server as soon as > the sipura sip device goes offhook and before > any digits are pressed. This way asterisk can > provide the dialtone and the dialplan. > This also allows me to play a greeting to the phone > before giving them a dialtone. > > Is there any way to do this, like possibly having the sipura > device dial a predefined extension when it goes offhook? SPA-841's can do that - look for 'hotline' in the admin guide. Or do you mean the SPA -1x, -2x, -3x stuff? Dunno bout those...since the SPA-841 hotline trick is just a special dialplan command, i think it might work on the SPA-[123]x models, but i havn't tried it yet. -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Current status of voicemail monitoring?
On Friday 13 May 2005 10:13 am, Nathan Pralle wrote: > Hi all. > > I'm curious as to the current status and development of a way to monitor > incoming voicemail in Asterisk. IE: The "screen calls with the > answering machine" feature -- the ability to listen to and break into a > currently-recording voicemail if you want to. > > This feature would be very helpful for our application. I've seen > various things in the archives about this but none where someone has > said, "Yeah, we have this, it works." Does anyone have a current > implementation of this and/or a reasonable alternative, and how did you > do it? > Hackish option -- See call come in using FOP (flash operator), watch it go to voicemail, use ZapBarge to listen to voicemail being recorded. If you want to talk to the caller, use FOP and drag the incomming Zap line onto your phone again, then answer your phone. See FOP docs on transfering calls. Hackish, but it works. I've used it myself several times. This, of course, assumes you are using Zap channels for incomming calls. If not, then you'd need to find another way to listen to incomming calls - perhaps ChanSpy, tho i've not been able to get that to work - crashes my * box with CVS HEAD. Anyway, let me know if thats helpful or not, and we can play around. Oh, another thought: One other way to do this (and this will be comples): 1. call comes in, extension is dialed. 2. Your ext macro (assuming you use a macro for extensions) first calls a script to add your phone to a dynamic meetme conf. 3. The macro then dials this call to the dynamic meet me conf. 4. The script refered to in '2' should me a manger script. The script could use standard manager API stuff like 'action: originate' to dial to your phone from the meetme conf. If your phone doesnt answer, the voicemail app would then play the audio and record from meetme instead of directly from the incomming line/ 5. If you wanted to listen to the caller recording VM at that point instead of answer the call, just dial into the dynamic meetme conf with your phone muted. This assumes the meetme conf was created with the 'q' option to disable announce of joins. 6. If you want to talk, then just unmute your phone. Only question is how to turn off VM. Does standard app_voicemail.o respond to DTMF to stop recording? Would DTMF work in the meetme to stop recording? Anyway, those are just the top-of-the-head thoughts. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Giving user progress in an voice menu system
On Thursday 12 May 2005 3:43 pm, Sean Kennedy wrote:
> Hi all,
>
> I have a voice menu system ( Outlined below ), and I'd like to give the
> user some feedback when they dial an extension ( ringing, music,
> SOMETHING ). As it stands, when a user enters an extension from the
> menu system, they hear silence while the line rings. I even tried
> including the Ringing application before calling my macro to dial the
> phones, with no luck.
>
> Any help is apprecaited.
>
Odd - my receptionist was having a similar problem. I used the stdexten macro
that came with the demo files - when ever someone dialed directly (inside) or
directly thru the IVR (no receptionist pickup) - the ringback was fine. But
when the receptionist picked up and transfered - no ringback. All three
methods of dialing went thru the stdexten macro - very puzzling. The solution
I finally came up with was to add the 'm' option to the 'Dial' command.
Code speaks louder than words, so here you go..its obviously modified a bit -
but all should be self explanitory. The "SIP/op" channel is our receptionist
phone. The macro only adds the MOH option if the call is from the
receptionist phone, otherwise it leaves all options at default.
Anybody else have any other solutions or need debug outputs to figure this
out?
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,GotoIf($["${CHANNEL:0:6}" = "SIP/op"]?999|1)
exten => s,n(dial),Dial(${ARG2},20,${opt})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => 999,1,SetVar(opt=m)
exten => 999,n,Background(transfer)
exten => 999,n,Goto(s,dial)
;exten => s-NOANSWER,1,Voicemail(u${ARG1})
exten => s-NOANSWER,1,AGI(vm.pl|u${ARG1})
exten => s-NOANSWER,2,Goto(${vm-exit-context},s,1)
;exten => s-BUSY,1,Voicemail(b${ARG1})
exten => s-BUSY,1,AGI(vm.pl|b${ARG1})
exten => s-BUSY,2,Goto(${vm-exit-context},s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => *,1,AGI(checkvm.pl|${ARG1})
exten => *,n,Goto(mainmenu-restart,s,1)
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] Sipura 841 and headset
On Tuesday 10 May 2005 9:45 am, David Masure wrote: > Hi folks ! > > I bought two sipura 841 phones. I used to have GN Netcom headset which > I connect instead of the handset. The problem is that I don't have any > sound coming out the headset and I can't speak neither ! > ... > > Orcan anyone advise me on headset working with the sipura 841 ? I just use a standard 2.5mm headset that plugs into the small port on the right side of the phone about 1.5" below the screen. The 2.5mm size plug is standard for cell phone headsets and cordless phone headsets, at least here in the USA - dont know about france. The 2.5mm headset works fine on the 10+ SPA-841's that I've used it on (several managers use headsets in my office with the SPA-841.) -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interconnecting two lans using Asterisk over a PSTN
On Tuesday 10 May 2005 8:50 am, Brice Muangkhot wrote: > Hello, > > I am a newbie in Asterisk IP PBX but I am very impressed by its > functionalities. > I have read that It can work over IP network and across the PSTN. > I am not very sure how it works over the PSTN.. > In case if people have not yet the Internet or SDL access, I would like > just to know if it is possible to interconnect two IP LANs using the > traditionnal analog Network and Asterisk PBX? In such I would like to send > voice and data in the same media and all my IP applications should work, it > is like the PPP IP link over PSTN I would like to have. > > If one have an Internet/SDL access is it possible to have a dual access, in > such if one network fails, I can switch automatically to another? > Here below the network architecture I imagine. > > (IP LAN1)-ASTERISK-(Digium FXO card)<-->(PSTN)<-->(Digium FXO > Card)-ASTERISK-(IP LAN 2) > I'm not sure _exactly_ what you mean...but wouldn't it be simpler just to use two modems to tie each lan together - maybe even have the modems in the Asterisk box and then use standard linux stuff to do the routing of the packets - no asterisk _needs_ to be involved - the packets could be iax trunks or whatever, but the actual bridging of the the lans doesn't need asterisk, just two modems and a NIC in the server on either end. Something like this: (IP LAN1)<->LINUX+MODEM<>(PSTN)<>LINUX+MODEM<->(IP LAN2) And on LAN1 or LAN2 could be an Asterisk server, or a web server, or whatever - the linux servers with the modems just do standard routing... At least, i think thats what your asking.. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New script: /usr/bin/asteriskdial + Kontact
Hey all - Attached is quick perl script i whipped together this morning to dial from Kontact (well, KAddressBook). You'll need Net::Telnet (perl -MCPAN -e 'install Net::Telnet') for it to work. You'll also need to make sure you've got a valid user/secret in /etc/asterisk/manager.conf on your * server for this to work. Usage: /usr/bin/asteriskdial --from --to "%N" ...assuming you've set the defaults correctly for your system in the script. To use, in Kontact, goto your contacts, click Settings, click Configure KAddressBook, goto the General Tab, and in the 'Script-Hooks' section in the 'Phone' box, put '/usr/bin/asteriskdial --from --to "%N"' KDE (well, kaddressbook) will then allow you to click on numbers in your contacts and substitute the %N for the number clicked. Assuming you've chmod'ed +x the script and set the server, user, and secret correctly, when you click on a contact's number, your phone will ring. Any questions, let me know! Cheers! -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 asteriskdial Description: Perl program ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
On Tuesday 03 May 2005 11:40 am, Ryan Courtnage wrote: > Hello all, > > Everyone has probably experienced this at some point in the past: > You pick up your analog phone. Rather than hearing dialtone, you are > connected with someone who has just called you. Neither you nor them > heard a ring. > > Maybe it's just me, but it seems these "freak incidents" would occur > more frequently years ago, than now. > > I've now experienced this a couple of times with an * system (TDM400p > - quad FXO): > A SIP exten dials digits which are answered by a Zap trunk. As soon > as Zap answers, the SIP extension is connected with an inbound (PSTN) > caller (who was expecting to hear an IVR). > > My questions are: Who's to blame (telco, tdm card, * config, > gremlins)? Is this avoidable? I dont know who to blame, but we've had the same problem here with our small sales team. The sales team (about once a week) will dial a call on their analog phones (analog cordless phones plugged into a few SPA-2001s) - they press 'talk', dial the #, then immediatly are connected to an incomming call... (I use two TDM quad FXO cards to service 8 incomming lines from Sprint). I havnt been able to track it down, and its not reproducable manually... Anybody have any ideas? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM users: modified zttest.c for testing
On Tuesday 03 May 2005 11:48 am, Rich Adamson wrote: > The design objective of the TDM (and x100p) cards was to transfer > 8,192 bytes of data from the card in exactly 1.0 seconds. > The above sample indicates my system required 1.023856 seconds to > accomplish this, or 23856 microseconds too late. > ... > > It would be very interesting to see everyone's results in running > this, and even more interesting to report the results with the OS > distro in use, mobo in use (if known), etc. If anyone actually > get's a result that is very close to 1.000 seconds, I'd really > like to know more about those systems. (email off list is fine > if you want.) > Results summary: Best: 1.024028 -- Worst: 1.023796 -- Average: 1.023886 Distro: FC3 [EMAIL PROTECTED] zaptel]# ./zttest-mod -v Objective: to read 8192 bytes from TDM card in 1.00 seconds. Opened pseudo zap interface, measuring accuracy... 8192 bytes in 1.023861 seconds 8192 bytes in 1.023878 seconds 8192 bytes in 1.023878 seconds 8192 bytes in 1.023894 seconds 8192 bytes in 1.023861 seconds 8192 bytes in 1.023878 seconds 8192 bytes in 1.023879 seconds 8192 bytes in 1.023878 seconds 8192 bytes in 1.023877 seconds 8192 bytes in 1.023912 seconds 8192 bytes in 1.023845 seconds 8192 bytes in 1.023879 seconds 8192 bytes in 1.023911 seconds 8192 bytes in 1.023898 seconds 8192 bytes in 1.024028 seconds 8192 bytes in 1.023796 seconds 8192 bytes in 1.023916 seconds 8192 bytes in 1.023887 seconds --- Results after 18 passes --- Best: 1.024028 -- Worst: 1.023796 -- Average: 1.023886 # uname -a Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25 EST 2004 i686 i686 i386 GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 1 model name : Intel(R) Pentium(R) 4 CPU 1.50GHz stepping: 2 cpu MHz : 1483.665 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 2924.54 -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the beginning of voice menu is cutted
On Friday 29 April 2005 12:12 pm, Kib Eki wrote:
> Hi,
>
> when I dial my voicemenu the menu voice is always cutted so that i only
> hear 'word from password.
> What do i have to configure so that i hear the full text from the
> beginning?
>
> thanks, Kib
You might try inserting a Wait in your menu ...e.g...
exten => s,1,Answer ; answer the channel
exten => s,n,Wait(2) ; give the channel time to initalize (2seconds)
exten => s,n,Background(some-recording)
The 'Wait' supposedly gives the channel time to 'initalize' and get ready to
send audio. If you start dumping audio ('Background') down a channel not
initalized, you wont hear anything until the channel is initalized, even if
the audio has already started.
At least, thats my non-developer-ish understanding of the sequence of events
after having the same problem myself...
HTH,
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] RE: Number of production asterisk systems (Christopher Jacob)
On Thursday 28 April 2005 1:52 pm, Greg Eaton wrote: ... > In terms of finding out true factual data, in terms of reference perhaps > digium would share market data (but then they might not!) and we could > extrapolate from there, but given how much of a community is building up > around the product I would suggest some form of "self-registration". > i.e. you/someone sets up a webpage and then users are encourage to > 'register' their systems on this web page, I guess with basic data such > as Country, Company, Industry, Location, Number of Sites, Number of > Users, usual daily call volume, gives us the basic metrics. Contact > information could be added for reference which might be of use. I just put together a small production survey form on a mysql database using the fields you mentioned and put it on my R&D server. I'd encourage everyone to do as Greg suggested and register your system install at: http://207.40.85.50/production_survey.cgi The data from the survey is freely available upon request. I'll try to put together a web interface to view the db when i have a few more free hours. Let me know if anyone has any trouble accessing the script or has any suggestions to make the survey better. Note: Its a non-mod_perl script (just 96 lines of quickly written perl code using HTML::Template) running on a MySQL database. It uses Randal Schwartz's technique to limit instances of CGI scripts from his 2000 column at http://www.stonehenge.com/merlyn/WebTechniques/col54.html. Anyway, all that just to say be nice and dont melt my serves. :-) -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prefix to CALLING Number ?
On Thursday 28 April 2005 11:07 am, barney wrote:
> Hi there,
>
> I`m trying to add some prefix before my local extensions, when my calls are
> routed to ZAP trunk.
>
> (i.e.: my local extension is , and i would like to send request to my
> telco provider with source phone number 55)
>
> Is there any way to do this ? I just know to add prefix (via prefix
> application) to the called number (but not calling).
>
Thread on this 2 days ago.
Serach the archives. See footer on every message in this list.
For those who dont want to google archives, here ya go:
exten => ,1,Dial(Zap/g1/5${EXTEN}/);
Just put the number to add before the number to dial:
For example, to dial XXX-XXX and put a '9w' before the number when sending to
a zap trunk:
exten => _NX,1,Dial(Zap/g1/9w${EXTEN})
Cheers!
-josiah
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Re: [Asterisk-Users] Redirect two channels to each other?
On Thursday 28 April 2005 12:57 pm, Nicolás Gudiño wrote: > > It almost sounds like there needs to me a new manager action: > > > > Action: Bridge > > ChannelA: SIP/199testfone-1f3c > > ChannelB: Zap/6-1 > > > > It sounds like the intrinsic functionality for 'bridging' is already > > there in Asterisk (duh!), it just needs to be encapsulated in a manager > > action. > > Yes, we need that action on the manager! (but replace ChannelA and > ChannelB to Channel1 and Channel2 as on the "link" event). Fine by me...but does anybody know the relevant subs in any of the asterisk source that actually does the bridging? I mean, is it possible that its as easy as: 'void ast_bridge_chan( ast_chan * a, ast_chan *b )' and we just need to package it as a manager action? Anybody have any pointers on how to proceed? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RJ45 to RJ11?
On Wednesday 27 April 2005 2:02 pm, Paul Shiflet wrote: > I just received my TDM400 card from digium with 2 fxo and 2 fxs > interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS > phones. How do i interface my POTS phones with this; can i just crimp an > RJ45 connection on the end of the phone cord? > Umm...can you just plug it in? -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redirect two channels to each other?
On Wednesday 27 April 2005 11:38 am, Alexander Lopez wrote: > As ny 10 year old step-daugher says "I don't get it.." > > Can't you just do a redirect if you specify the channels, * doesn't care > if they are bridged together or not. You may end up with zombie > channels if the other leg does not drop, but you could do a soft hangup > and take care of that.. > > > Or am I missing something I dunno...maybe _im_ missing something.. IIRC (http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Redirect), Manager action 'Redirect' only takes Channel, ExtraChannel, Exten, Context, and Priority as parameters. Example (transferring existing 2 party call to a meetme room): Action: Redirect Channel: Zap/73-1 ExtraChannel: SIP/199testphone-1f3c Exten: 8600029 Context: default Priority: 1 The problem is that you cant redirect to an existing _channel_. I dont know of any channel 'hack' like there is for Local extensions (e.g. to make an extension look like a channel, use Local/[EMAIL PROTECTED], etc - is there the inverse of that? Make a channel look like an extension?) It almost sounds like there needs to me a new manager action: Action: Bridge ChannelA: SIP/199testfone-1f3c ChannelB: Zap/6-1 It sounds like the intrinsic functionality for 'bridging' is already there in Asterisk (duh!), it just needs to be encapsulated in a manager action. Any takers? Maybe a bounty is needed...? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redirect two channels to each other?
On Wednesday 27 April 2005 10:40 am, Tony Mountifield wrote: > I've been scratching my head trying to think of a way to do this, but > without success yet. > > I'm using the Manager API. If I have two channels linked to each other > (i.e. direct connection), or even if they are independent channels, > I can transfer them both to the same extension by using Action: Redirect > and using Channel: for one and ExtraChannel: for the other. This is most > useful for putting them both in the same Meetme conference. > > What I want to do is find a way to take two unrelated existing channels > (which for the sake of argument might be sitting in MusicOnHold, separate > conferences, the same conference or whatever), and link them together > into a direct call rather than having them talk via their own Meetme > conference. I have no ideas, other than Meetme. It sounds like it would involve some direct modification of the * code. Its similar to the pickup code - perhaps start there. Let me know if you find anything - id be intersted in a solution for it as well, i just dont have the time to find a solution. -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US$100 bounty for two features in voicemail
On Tuesday 26 April 2005 8:45 pm, Richard wrote: > Hi All, > > This is a US$100 bounty for two features in voicemail. > > . need it by May 1st. > . feature 1, one new option to play message from latest to the oldest. Now > it only supports playing from oldest to latest. I'd like this as an option > for each mailbox. Therefore some users have latest first while others have > oldest first. > . feature 2, simplified folder, only two folders -- new and old. Now it > supports several folders. Some users are overwhelmed by this. Most > voicemail system only supports two folders new and old. When they save, it > goes to old folder. Again, I'd like this as an option for each mailbox. > . open source and we will release the code to public > . I'd expect there will be code modification in apps/app_voicemail.c. It > should be based on the latest cvs stable version. > I didn't lile the logic of the builtin voicemail system so I rewrote the entire voicemail system in Perl for my company. It's worked well with over 50 users thusfar. It works with the asterisk MWI as well. I can easily modify it to support these features and have it ready by May1st if Perl is acceptable for this bounty. Since this is AGI, it will work with the latest cvs stable version. **Please CC me directly on a reply to this, since I sometimes miss replies on the list** -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAC and IPs
On Tuesday 26 April 2005 8:01 pm, William Suffill wrote: > Why not DBGet from the SIP.Registry in ASTDB? Wouldn't that be cleaner > since it would only return the 1 you want instead of parsing what > could be a load of sip peers? Hmm, great idea! (This is one of those "DUH!" moments for me) - I never thought of trying the ASTDB - thanks for the reminder. Yes, that would be much easier than parsing sip show peers. Just use one of the dialplan commands to cut the return value at the first ':' and there ya go! Google 'site:voip-info.org Asterisk variables' or something like that for string maniupluation. -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAC and IPs
On Tuesday 26 April 2005 1:04 pm, Anton Krall wrote: > Could you send it to me offlist? This sounds like a good approach.. > > Know if you can make a bash script an agi? You can use any language you want to do AGI. http://www.voip-info.org/wiki-Asterisk+AGI I'll send you the regular perl script off list. -josiah > > |-Original Message- > |From: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] On Behalf Of > |Josiah Bryan > |Sent: Martes, 26 de Abril de 2005 09:59 a.m. > |To: Asterisk Users Mailing List - Non-Commercial Discussion > |Subject: Re: [Asterisk-Users] YAC and IPs > | > |On Tuesday 26 April 2005 9:58 am, Anton Krall wrote: > |> Guys. > |> > |> Im using YAC to send callerid info to PCs and I was > | > |wondering if there > | > |> is a way to get the IP of a certain SIP or IAX > | > |client/technology when > | > |> a dial command is issued. > |> > |> For example, if the dialplan has a dial sip/client or > | > |iax2/client, is > | > |> there a way to get the current clients IP so I can pass the > | > |parameters > | > |> to the system call that send the YAC callerid info? > | > |Simplest way probably would be to parse the output of 'sip > |show peers' or a similar IAX CLI command (I dont use IAX, so I > |dunno.) I've got a small perl script that parses 'sip show > |peers' to get the peer name (SIP/whatever) and the matching IP > |address - just a simple regex exercise, really. It could > |easily be converted to AGI where one could call: > | > |exten => s,1,AGI(tech2ip.pl|SIP/whatever) ; IP is put in TECH_IP > | > |If anybody wants the script, let me know. > | > |Cheers! > |-josiah > |___ > |Asterisk-Users mailing list > |Asterisk-Users@lists.digium.com > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YAC and IPs
On Tuesday 26 April 2005 9:58 am, Anton Krall wrote: > Guys. > > Im using YAC to send callerid info to PCs and I was wondering if there is a > way to get the IP of a certain SIP or IAX client/technology when a dial > command is issued. > > For example, if the dialplan has a dial sip/client or iax2/client, is there > a way to get the current clients IP so I can pass the parameters to the > system call that send the YAC callerid info? Simplest way probably would be to parse the output of 'sip show peers' or a similar IAX CLI command (I dont use IAX, so I dunno.) I've got a small perl script that parses 'sip show peers' to get the peer name (SIP/whatever) and the matching IP address - just a simple regex exercise, really. It could easily be converted to AGI where one could call: exten => s,1,AGI(tech2ip.pl|SIP/whatever) ; IP is put in TECH_IP If anybody wants the script, let me know. Cheers! -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan - How to prepend a digit
On Monday 25 April 2005 3:05 pm, Daniel Salama wrote:
> I'd like to create a dial rule that when someone tries to dial a
> particular number, the same number is dialed, except that prefixed with
> some additional digit(s). How can this be specified on extensions.conf?
exten => 1234,1,Dial(Zap/g1/555${EXTEN})
Just put the extra digits before the extension to dial on the device.
-josiah
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Re: [Asterisk-Users] Wait in Dial String
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
> Dear All,
>
> My boss has placed a requirement for me to forward all our IDD calls
> through a partner's IDD service, which requires us to call a 8 digit
> number, wait for 1 sec, before we send in the foreign number we're trying
> to call.
>
> As I can't find anything on getting the PBX to wait, i'm only removing the
> 1st 3 digits (900) and sending in an extra 1 to simulate the wait. It
> works, but not all the time. Is there anyway that I can place a wait
> command here? I'm tried placing w / p but both don't works. Would like to
> seek your kind assistance!
>
> exten = _9001.,1,Dial(Zap/g1/64919669,,D(${EXTEN:3}),)
> exten = _9001.,n,Hangup()
>
Try 'w',
E.g. for my old bridge to BizFon, I had to dial 9, wait, then the number:
exten => _NX,1,Dial(Zap/g1/9w${EXTEN})
Just put the 'w' between the numbers that you want it to 'wait' at.
-josiah
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IT Coordinator
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Re: [Asterisk-Users] DIAL FROM CONSOLE
On Monday 18 April 2005 5:56 pm, Dan Levine wrote: > I forgot the command to have asterisk dial and hangup from the console. dial hangup (try 'help' from the CLI) -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Steal a call from a SIP extension
On Thursday 14 April 2005 10:24 am, Sean Kennedy wrote: > Hi all, > > I think I've seen this somewhere, but I can't remember where; Is it > possible to steal a call from a sip extension? Let me explain what we > are trying to do: > > Parking calls is a good thing, but having to remember an extension may > be a bit much to ask my user base who is used to seeing line presences > on their phones ( old avaya partner ACS system ). I'm thinking they'll > keep forgetting what extension a call is parked on. I would like for > them to be able to put a call on hold on their extension, and have > someone else be able to "steal" it off that extension from a different > extension. > > Is this possible? If so, can I get the terminology for this ( I can do > my own research if needed )? Should be an incredibly easy modification of a script I wrote to do call pickups. Several people have tested this script out and it seems to be working at other * installs. In house, we use it heavily with about 40-50 users to pickup ringing calls. The script would just need to be modified to not check if an extension is ringing - just check if the extension is 'live' - e.g. a channel exists for this exension. Once the logic of finding out if an ext is live and all the AGI stuff to get an extension via DTMF is done, its very simple just to issue a Manager 'Redirect' command to redirect a call from one device to another. The key is to redirect the _calling_ channel instead of the _called_ channel - e.g. if a call came in on Zap/4-1 and called SIP/whatever, the Manager Redirect is issued to Redirect 'Channel: Zap/4-1' instead of SIP/whatever. Anyway, that is all handled by pickup.pl anyway - figuring out the channel to redirect, etc. If this sounds helpful or useful, let me know... HTH & Cheers! -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dumb question ?
On Tuesday 12 April 2005 12:45 pm, mr. barker wrote:
> Here it is
>
> exten => s,1,answer
> exten => s,2,SetCIDName('PMG')
>
>
> In a lot of config files I see "exten => s,"snip ..
> Is "s" just an extension or system variable for all extensions ? or
> something else ?
http://www.voip-info.org/wiki-Asterisk+s+extension
http://www.voip-info.org/wiki-Asterisk+standard+extensions
http://www.google.com/search?hl=en&lr=&c2coff=1&q=site%3Avoip-info.org+Asterisk+s+extension&btnG=Search
Google is our friend...
:-)
>
> Thanks
Your welcome.
Cheers!
-josiah
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IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] Multiple Servers and 1 Central Voicemail
On Tuesday 12 April 2005 11:49 am, Luki wrote: > > Also, what happens if for example, the user is accessing his VMB > > on server 1 and changes his password, then travel to where server > > 2 is and tries to access his VMB? the config on server2 would > > still have the old one so you need to sync voicemail.conf on > > all servers too ... > > If you use the realtime config via a DB, it should be OK. But I still > don't think that MWI will work properly if a message is left on server > A and user is actually registered on server B, which is NOT on the > same network and hence does NOT share the same voice mail spool. How > will B know there is a message left on A for the same user? Does > realtime share this info too? And if so, how does the message get > retrieved if B does not have access to files on server A, where the > actual message is? Why not just NFS mount the /var/spool/asterisk/voicemail directory from a central server? That way, all servers share the same spool and the MWI will get reflected on all servers. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple line usage on Polycom IP300
On Tuesday 12 April 2005 11:00 am, MobilPete wrote: > we tried both, setting it as same and also seperate. but niether worked. I've never used the IP300, but I do have an IP500 on our network. It has 3 line buttons, each line can do 2 simultaneous calls. Each line button registers as its own SIP device (op-1, op-2, and op-3). I wrote an AGI script to dial the IP500. It uses the * Manager to do 'show channels' to find the line button on the IP500 with the least number of simultaneous calls (e.g. which SIP device [SIP/op-1, SIP/op-2, or SIP/op-3]) then the AGI script just redirects the call to the next available line on the IP500 using AGI 'EXEC' to run the 'Dial' app. If anybody is interested in the script, ill try to clean it up enough to post. -josiah > ----- Original Message - > From: "Josiah Bryan" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, April 12, 2005 9:41 AM > Subject: Re: [Asterisk-Users] multiple line usage on Polycom IP300 > > > On Tuesday 12 April 2005 10:18 am, MobilPete wrote: > >> can anyone help ?? > >> trying to get Polycom IP300 to utilize both lines, would like calls to > >> roll > >> to open line when incoming call arrives while user is on line 1. Looked > >> everywhere and tried many things with no luck. > > > > Do you have your lines register sepratly? E.g. is there a seperate entry > > in > > sip.conf for each line or do they both register as the same sip device? > > > > > > > > -- > > Josiah Bryan > > IT Coordinator > > Productive Concepts, Inc. > > [EMAIL PROTECTED] > > (765) 964-6009, ext. 224 > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple line usage on Polycom IP300
On Tuesday 12 April 2005 10:18 am, MobilPete wrote: > can anyone help ?? > trying to get Polycom IP300 to utilize both lines, would like calls to roll > to open line when incoming call arrives while user is on line 1. Looked > everywhere and tried many things with no luck. Do you have your lines register sepratly? E.g. is there a seperate entry in sip.conf for each line or do they both register as the same sip device? -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manipulate Asterisk Database from manager?
On Monday 11 April 2005 2:24 pm, Matt wrote: > Ahh dbput probably will do what I am looking for.. thanks! > > On Apr 11, 2005 2:23 PM, Brian Roy <[EMAIL PROTECTED]> wrote: > > On Apr 11, 2005 10:16 AM, Matt <[EMAIL PROTECTED]> wrote: > > > Hi, > > > Is there anyway to manipulate the asterisk internal database from the > > > manager (the one you can telnet to)? And if so.. how does one do it? > > > (ie for enabling call forwarding, etc) > > > > Not that I'm aware of. You can do a 'listcommands' from the manager to > > verify everything that is avaiable to you. > > > > You can use the asterisk -rx "dbput" from a shell script if that > > suites your needs. > > Or: Action: Command Command: database put path/to key value -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reply-To?
On Friday 08 April 2005 1:12 pm, Bruno Hertz wrote: > John Novack <[EMAIL PROTECTED]> writes: > > Bruno Hertz wrote: > > > > <>Jean-Michel Hiver <[EMAIL PROTECTED]> writes: > > Jean-Michel Hiver wrote: > > Oops, sorry for the list reply :/ > > > > <>Actually, why does the Reply-To point to the Asterisk Users > > mailing > > list? This breaks the reply to sender only / reply to all / > > list reply > > functionality of my mailer. It's really broken :( > > > > Some would say your mail client is broken. What you're complaining > > about is generally called 'reply-to munging', and there's been a long > > discussion about this. Google reveals more, like these two oppositional > > opinions > > > > http://www.unicom.com/pw/reply-to-harmful.html > > http://www.metasystema.net/essays/reply-to.mhtml > > > > Regards, Bruno. > > > > > > And there probably will NEVER ba an agreement on this subject. > > Another list I am on even went so far as to take a poll, and it was split > > right down the middle, half taking the correct position outlined in the > > first article, and half the second, much less flexible, position.. > > > > The really curious thing on this list is every so often, if I choose to > > reply, the poster AND the list appear, but mostly just the list, as if > > the poster had some control as well. > > Well, the reason for the latter apparently is that, in some postings to > this list, there's actually two entries in the reply-to header, the posters > mail and the list address, while in others it's only the list. Why this > happens is above me, though, I thought it should be either/or. Though it may be 'technically' correct per RFC guidlines, is it really correct usage-wise? Commen sense tells me that when I click reply, i want to reply to the message, and i want the message to go back to where it came from, in this case the mailing list, not the individual. The individual sent it to the *Mailing List*, not to *Me*. The *Mailing List* then sent it to me, therefore I am replying to the *Maling List*, not the individual. Does that make sense? Yes, RFCs may say different, but are they really logical to the common man? Or even to technical users who dont care about the RFCs and just want to do their work? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inquire about connected channel (show channels)
On Friday 08 April 2005 10:04 am, Jerry Geis wrote: > is there an easier way to ask through the manager api > what the connected channel is for a given channel. > > Example: I dont know the session number for SIP/401 > but I what to know what channel SIP/401 is connected to. > > SIP/401 is presently something like SIP/401- type session number > and the response to the this command would be > SIP/422- > > where SIP/422- is the channel and session information that SIP/401 > is connected to. > > I know this information can be parsed out of "show channels" > but I was just wondering if the is an easier way? Its rather simple with AGI + Asterisk Manager interface. I wrote a little AGI (perl) script that connects to * and parsers 'show channel X' and grabs the 'Direct Bridge' line for that channel. This would give you, say, SIP/422-xxx as the Direct Bridge for SIP/401-xxx. I use this for transfering calls for my receptionist. So, to answer your question, just parse the output of 'show channel SIP/401-' and grab the 'Direct Bridge' line. Thats about the easiest that I know of.. HTH - -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice controlled calling?
On Thursday 07 April 2005 4:43 pm, Andrejus Stavickis wrote: > Hi, > > As far as I know, there is a possibility to make a phone to dial an > extention when you pickup the phone. If the phone can do that (I know > the Grandstream BT101 has that possibility I have no exposure to other > phones yet) then you can do whatever you wish within the extension user > gets in. > The SPA-841 can do hotline calling - special dialplan string (see the admin docs) - very useful for doing BizFon-like auto-voicemail checking. The OP prompted me to lookup Sphinx - im installing it as i type. Sounds like an interesting idea(*mad scientist laugh*.muahahaha... :-) -josiah > Sincerely, > > --Andy > x6722 > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of magnus > > Sent: Thursday, April 07, 2005 4:35 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Voice controlled calling? > > > > Hello all, rumours reach me of a way that the UK incumbent > > operator is planning to compete with VOIP by offering voice > > activated dialling, e.g. > > pick up the handset and through speech dial from your > > personnel directory, this leads me to wonder if this could be > > performed with Asterisk and Festival? I have looked in the > > WiKi and goggled, but can find no information on if this is > > possible, (particularly with SIP?) hence this question, has > > anyone achieved this? > > Intent would be to make is simple for non technical person - > > E.g. Grandma picks up the phone, does not have to worry > > about entering any digits and then makes call by voice > > control - for example "call daughter" etc. The key here is > > not to need any human interaction with the phone, other then > > picking up handset, the rest controlled by voice. > > Many thanks > > Magnus > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > _______ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Database lookups?
On Thursday 07 April 2005 3:19 pm, Jan Johansson wrote:
> Is it possible (How complicated is it?) to do this;
>
> Call comes in, connects to IVR
>
> IVR plays the usual "please type your order number, finish with pound"
>
> Then I would like to query a MSSQL database server, looking up the "Status"
> column from a row where ordernr = the entered order number.
>
> Depending on the result of the lookup, play one of two messages ("Yes,
> ready for pickup" or "No, your order is not ready").
>
> Can someone clue me in on which docs I should start with? Or is there an
> example of this somewhere?
Very simple. AGI. Write a custom AGI program (any language) to do the MSSQL
query. AGI can control playback of audio, etc. See
http://www.voip-info.org/wiki-Asterisk+AGI.
Cheers & HTH -
-josiah
--
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IT Coordinator
Productive Concepts, Inc.
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Re: [Asterisk-Users] Call Interception
On Thursday 07 April 2005 11:20 am, Rich Adamson wrote: > > > What you are asking for (in US terms) is directed call pickup. > > > > > > Asterisk does not have a directed call pickup implemented > > > within it. Not sure how one would try to implement that, but > > > a guess would be that it would require an external script > > > or app of some sort. > > > > Actually I've just tested this. I'm using * 1.0.7 with bristuff. > > There is an application PickupChan: > > > > *CLI> show application PickupChan > > *CLI> > > -= Info about application 'PickupChan' =- > > > > [Synopsis]: > > Channel independent call pickup. > > > > [Description]: > > PickupChan(Technology/resource[&Technology2/resource2...]): Tries to > > pickup the first ringing channel in the parameter list. > > > > It's working fine, although I'm not sure if it comes with asterisk or > > with bristuff ... > > What the OP was asking about was... if * phone x123 and x234 are both > ringing, how can he remotely pickup the call from x123 _only_. pickup.pl - I'm trying to get into a form that is world-consumable. Its a little AGI script that I made that uses the Manager interface to Redirect ringing calls based on user selection. Would that be something like what you're looking for or no? -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Interception
On Thursday 07 April 2005 9:14 am, cereal killer wrote: > > Yes. > > > > http://www.voip-info.org/wiki-Channels+and+Groups > > "A channel that belongs to a pickupgroup, can pickup > > all incoming > > > calls on the same callgroup by hitting *8" > > Thanks answering me, that works with the *8 (and *02 > th e pattern in my company works too) but there is a > problem : how do you select the phone ringing to > pickup ? For example phones 23 and 24 are ringing ; > I'm 25 (same pickupgroup as 23 and 24 callgroup), How > do I decide either to take the 23 or 24 ? Seems the *8 > takes the first arrived call. Any idea ? > There is no way to do that (that I know of) in the default Asterisk setup. Which is I wrote a little Perl AGI script that lets users dial 200 to pickup a call. (Dial 200, then dial the extension at the prompt. The users phone then rings, with caller ID on the screen.) This works for any ringing channel on Asterisk, regardless of callgroup or pickupgroup. I suppose that could be added to 'limit' users, but its currently not implemented. You can pickup any channel that is ringing (SIP, Zap, etc.) with this script, since it just issues a Manager 'Redirect' action. Usage: exten => 200,1,AGI(pickup.pl) If anyone is interested in pickup.pl, let me know and I'll see what I can do to make it available. Cheers! -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk sounds
On Tuesday 05 April 2005 3:28 pm, David John Walsh wrote: > Dov, > > If anyone responds to your request privately, I'd apreciate it if you > were to forward it to me, as I need to translate them into several > european launguages. Guys - As others more enlightened than myself pointed out - Look at /usr/src/asterisk/sounds.txt, where /usr/src is the location of your asterisk CVS tree. sounds.txt has both the file name and the transcript of the audio. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI call problem
On Tuesday 05 April 2005 3:24 pm, Adam Robins wrote:
> I am issuing an AGI call in extensions.conf as follows:
>
> exten => 2000,1,Answer
> exten => 2000,n,AGI(script.pl)
> exten => 2000,n,Hangup
>
> Asterisk perl is installed. "script.pl" is a valid perl script in
> /var/lib/asterisk/agi-bin
>
> Output is:
>
> -- Executing Answer("SIP/2034-e908", "") in new stack
> -- Executing AGI("SIP/2034-e908", "script.pl") in new stack
> -- Launched AGI Script /var/lib/asterisk/agi-bin/script.pl
> -- AGI Script script.pl completed, returning 0
> -- Executing Hangup("SIP/2034-e908", "") in new stack
> == Spawn extension (intl-access, 2000, 3) exited non-zero on
> 'SIP/2034-e908'
>
> The problem is that the logic in the script is not executing.
>
Check the following:
-The script is executable (chmod +x /var/lib/asterisk/agi-bin/script.pl)
-The script has the right interpreter in the head. (#!/bin/perl or whatever
`which perl` gives you - put this on the first line of script.pl)
-If that doesnt work, run 'perl script.pl' from a shell on your asterisk box
and check for syntax errors. (Make sure to have 'use strict;' in script.pl).
-If no syntax errors, try stepping through it by issuing AGI result codes to
the script if that what your script is expecting (e.g. "200 result=0" or
whatever the rest of the codes are) and check for undefined subs or
something.
It helps to 'use strict;' in your AGI scripts so that the interpreter does
variable name checking for you and it shows up when you run your script from
the command line.
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"
On Tuesday 05 April 2005 2:18 pm, David John Walsh wrote:
> Hi all,
>
> For an event we are doing, we have been donated several analogue PSTN
> lines and an 8 port FXO bridge.
>
> On the bridge, we have set up each of the ports to work on the SIP
> protocol, and have referenced them, line1, line2, line3 etc for their
> username / password.
>
> I have placed the config in sip.conf, and they all work fine, inbound
> and out - for testing anyway!
>
> How do I get asterisk, to treat these 8 lines as one 8 call limit
> trunk? From a users perspective, all he/she needs to dial is
> 9 (where x's the number) to get any of the 8 outside lines?
>
> Sure I could "hardcode" somthing in each part of the extensions.conf,
> but if this trial is sucsessful, the number of lines may increase, and
> it would be nice to define the array once as it were.
>
I did something quite similar for my receptionist and her Polycom IP500 phone.
The IP500 can register 3 SIP lines - e.g. in sip.conf i have [op-1], [op-2],
and [op-3] defined. Now, to dial the operator I could just to
Dial(SIP/op-1&SIP/op-2&SIP/op-3), etc. - but that makes it look like there
are 2 missed calls for every call answered.
Therefore, I did something exactly like what you are trying to do - I
"trunked" the three SIP lines into one - that way, whenever somone dials
'204' for the operator, the system looks at the three sip devices (op-[1-3])
and dials the first available device.
Here's the relevant exten:
exten => 204,AGI(opdial.pl)
opdial.pl defines an array of operator lines (my @operators =
('SIP/op-1', ...)) and then opdial.pl connects using the manager API and
finds the first available operator channel from the list provided, then calls
$AGI->exec('Dial',$open_device);
If anyone wants more info, let me know and perhaps I can clean up the
opdial.pl enough for general consumption.
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] asterisk sounds
On Tuesday 05 April 2005 2:40 pm, MF Hulber wrote: > asterisk/sounds.txt > > Josiah Bryan wrote: > >On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote: > >>Hello all, > >> > >>I am looking for a list of all available sound files for asterisk and a > >>transcription of their content, so that I can have someone translate them > >>into portuguese. > > > >I vaguely remeber reading some file in my server that had a list of all > > the sound files and their transcripts...i just spent about 20 minutes > > looking for it in the /usr/src/asterisk CVS tree that I checked out - > > cant seem to find it off hand. Any body have any idea what that file is? > > > >-josiah LOL. I really do know how to grep. ...I think. Sorry guys..it must be monday...somehwere in the world. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk sounds
On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote: > Hello all, > > I am looking for a list of all available sound files for asterisk and a > transcription of their content, so that I can have someone translate them > into portuguese. I vaguely remeber reading some file in my server that had a list of all the sound files and their transcripts...i just spent about 20 minutes looking for it in the /usr/src/asterisk CVS tree that I checked out - cant seem to find it off hand. Any body have any idea what that file is? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple PBXs on one server.
On Tuesday 05 April 2005 1:24 pm, Scott wrote: > Is it possible to run more than one Asterisk PBX on a single server? > I don't think there would be a hardware restriction using modern gear > but is there limitations on installs etc? I know it would be trivial > to make multiple databases for AMP and likely use different ports for > the SIP proxy. > It sounds like it would be easier to just configure seperate contexts for everything - e.g. a seperate context for the 'other' PBX in extensions.conf - perhaps use include => to keep the dialplans seperate just for the sake of editing.. Anyway, that sounds easiest to me -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I retrieve voice mail in Asterisk
On Tuesday 05 April 2005 11:09 am, Chuck Bunn wrote: > Hi, > > I guess I am dense or something but I cannot figure out how to retrieve > voicemail using a SIP SJPhone or and Analog phone with Astyerisk. I > googled (voicemail +retreive :lists.digium.com) and did not get much. > Everything works. I can ring each extension and if it doesn't answer it > goes to voice mail, but I can't figure out how to retrieve it. > Very simple. You need to set up a VoicemailMain extension in your extensions.conf Example: exten => 8500,1,VoicemailMain Infact, I believe that is in the sample extensions.conf in the initial distribution. Then just dial 8500 from your phone, enter the extension as defined in voicemail.conf and password for that extension (extension is what you put in when Allison asks for 'mailbox...'). That help at all? -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call redirection from outside line?
On Monday 04 April 2005 7:43 pm, Adam Goryachev wrote:
> On Mon, 2005-04-04 at 16:44 -0500, Scott Nelson wrote:
> > On Apr 4, 2005, at 3:04 PM, Jacob Cazzell wrote:
> > > I looked around but I can't quite figure this configuration out. I
> > > would like the ability to allow a user to call in to a number and be
> > > able to transfer back out after entering a "passcode" (to prevent just
> > > anyone from making calls through my system).
> >
> > That is the DISA application. See
> > http://www.voip-info.org/wiki-Asterisk+cmd+DISA
>
> Or for much tighter control on destination numbers they can call, you
> could just drop them into a context that has various extensions
> configured which will restrict them to dialling a 'local number' or
> whatever you prefer
>
> PS, this might also be possible with DISA, never looked, but there is
> always more than one way to skin a cat :)
Or, for much more flexible control of authentication, you could use a custom
AGI script + 'fake' DISA context (e.g. exten => s,1,Background(dialtone))..
Which is what i did for our shop.. wrote a simple Perl script that
authenticates the users:
[login]
exten => _7.,1,AGI(login.pl|${EXTEN:1})
You could also use login.pl sans the arg, which would cause login.pl to prompt
for extension using 'vm-extension'. Upon successful authentication, login.pl
redirects the call to 'internal,s,1' (or whatever you want to call it.) My
'internal' context looks something like this:
[internal]
exten => s,1,Background(dialtone)
include => extensions
include => local
include => longdistance
exten => t,1,Goto(s,1,)
exten => i,1,Playback(pbx-invalid)
exten => i,n,Goto(s,1,)
The timeout logic is a bit more complex (counts the number of timeouts and
hangs up after 2 timeouts, etc.) - same with the 'invalid' logic...but you
get the idea.
If anybody wants more info, feel free to ask. I also have the dialtone.gsm
file available - or its easy enough to record your own.
Cheers!
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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Re: [Asterisk-Users] Supervised transfer problems
On Monday 04 April 2005 6:23 am, Daniele Gallina - 3P System S.r.l. wrote:
> Hi all, when I try to transfer a call asterisk say me:
>
> -- Executing SetCallerID("SIP/20012-cb87", ""Gallina Daniele"
> <20012>") in new stack
> -- Executing Dial("SIP/20012-cb87", "SIP/20013") in new stack
> -- Called 20013
> -- SIP/20013-034d is ringing
> -- SIP/20013-034d answered SIP/20012-cb87
> -- Attempting native bridge of SIP/20012-cb87 and SIP/20013-034d
> -- Started music on hold, class '3psystem', on SIP/20013-034d
> Apr 4 12:12:39 NOTICE[4710]: chan_sip.c:5161 get_refer_info: Supervised
> transfer requested, but unable to find callid
> '[EMAIL PROTECTED]'
I've been having the same 'problem' with my Polycom SoundPoint IP500 phone.
When our receptionist hits transfer, , transfer - then I see a notice
just as above. The odd thing is that the call ID is given as on the phone,
not on the server. E.g. the IP in ''[EMAIL PROTECTED]" is the IP
of the Polycom phone, not the * box. Is there any way to fix this? Rewrite
sip headers? Any ideas?
-josiah
--
Josiah Bryan
IT Coordinator
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224
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