[asterisk-users] How to resolve CallerID from AudioCodes FXO
Hi, I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming outgoing calls. However, I noticed that the caller ID of the caller coming from the FXO displays its endpoints assigned number and not the actual caller's ID coming from PSTN. Hope someone is using the same scenario and could share on how to resolve the caller ID/Number. Thanks. Angel - Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why there havn't app_meetme.so file about asterisk1.4.0?
asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout in IAX vs SIP
On 02/01/07 02:15 Olle E Johansson said the following: both channels should act the same unless there's a configuration that's giving wrong information to chan_sip, like you having a username= or defaultip= setting. how does a username= entry in sip.conf affect dialling behaviour when the phone is not registered ? by default as a matter of practice, we have username=something for our peers, though they may be on dynamic IP addresses and register with asterisk. is what we're doing a Bad Thing(tm) ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get the status of failed call files
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and channel_status doesn't seem to be any good.
thanks in advance.
the event you received for OriginateFailure has a 'Reason: ' code.
that code breaks down as
0 = UNKNOWN FAILURE or DISCONNECT
3 = AST_CONTROL_RINGING (no answer)
5 = AST_CONTROL_BUSY
1 = AST_CONTROL_HANGUP
8 = AST_CONTROL_CONGESTION
i didn't originate the call with manager, but with a call file, so i can't
get manager events. i there any other way of finding this out?
--
- Rich Doughty
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[asterisk-users] extensions.conf gotoif and label
Hello,
I got a little interogation about these 3 points.
I want to write something like this sample in my extension.conf. I have
tested and it works but I don't know if it is a good way to make a menu.
I don't want to put number as it is boring to maintain.
Does anyone know if there is some problem to write like this?
exten = 7890,1,Wait(1)
exten = 7890,n(lbl0),Read(REP|annonce|1)
exten = 7890,n,GotoIf($[${REP} = 1 ] ?lbl1:lbl2)
exten = 7890,n(lbl1),noop( hit 1 ! )
exten = 7890,n,system(echo you hit one)
exten = 7890,n,Hangup
exten = 7890,n(lbl2),GotoIf($[${REP} != 2 ] ?lbl3)
exten = 7890,n,noop( hit 2 ! )
exten = 7890,n,system(echo you hit two)
exten = 7890,n,Hangup
exten = 7890,n(lbl3),noop( hit something else ! )
exten = 7890,n,system(echo you hit another key)
exten = 7890,n,goto(lbl0)
Another question on the web i have seen there is some trouble using
while application. Is that still true or it was an old release wich
get this problem?
Thanks ,
Nicolas
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[asterisk-users] Enhanced PickupChan
Hi All,
I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp .
from extensions.conf:
exten = 0,1,Dial(SIP/eosoiris|20|tTrR)
exten = 200,1,Dial(SIP/dzalewski|20|tTrR)
exten = _7.,1,Pickup2(${EXTEN:1})
When I try to pickup ringing SIP channel from other IP headset I go
disconnected.
here is debug from asterisk CLI:
-- Executing Dial(SIP/eosoiris-081b5e40, SIP/dzalewski|15|tTrR) in new
stack
-- Called dzalewski
-- SIP/dzalewski-081bb380 is ringing
-- Executing PickUp2(SIP/kitchen-081c08c0, 200) in new stack
find_matching_channel: pattern='200' state=5
find_matching_channel: trying channel='SIP/kitchen-081c08c0' state=4
pattern='200'
find_matching_channel: trying channel='SIP/dzalewski-081bb380'
state=5 pattern='200'
find_matching_channel: trying channel='SIP/eosoiris-081b5e40' state=4
pattern='200'
find_matching_channel: trying channel='Zap/3-1' state=6 pattern='200'
find_matching_channel: trying channel='SIP/developers-0819bca0'
state=6 pattern='200'
== Auto fallthrough, channel 'SIP/kitchen-081c08c0' status is 'UNKNOWN'
Any ideas?
Thank you in advance,
Dominik
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[asterisk-users] CDR - uniqueid
Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan programming vs. AGI vs. ???
This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that hook into our CRM and Accounting systems for fault reporting and credit card payments etc. What you need is the tool for the job like everything in life :-) Jon Farmer Telford, Shropshire, UK - Original Message From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 31 January, 2007 12:20:11 AM Subject: [asterisk-users] Dialplan programming vs. AGI vs. ??? Just a general question on dialplan programming... I've implemented a fairly full-featured system using dialplan code only. I've not used any AGI for it, yet it ticks all the boxes I want it to tick (diverts, follow-me, voicemail, dnd, outdialing restrictions, simple auto-attendant, and numerous star codes to control it all) This is all aimed at the small/medium office PBX type application. But I'm curious as to the approach others use. Is doing dialplan coding in an AGI more efficient, or do people just do it that way because it's easier than learning dialplan code? Or are there some things that people think they can't do any other way? So I'm just after some ideas, really, possibly to work out if it's worth my while going down the AGI route for future projects, or not!?! Any feedback is most welcome! Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: why there havn't app_meetme.so file aboutasterisk1.4.0?
You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org Àî¾ý [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI script callerid question
Have you tried
phpagi
http://phpagi.sourceforge.net/
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Michelle Dupuis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 26 January, 2007 5:52:27 PM
Subject: [asterisk-users] PHP AGI script callerid question
I am trying to set
callerid from a PHP script, using one of two functions as shown below (setid1
and setid2). The first function works great with regular names and
numbers, BUT, if I call the function with (Test,UnknownNumber), the cid
number gets set to asterisk. Why is my passed number parameter not being
accepted in this case?
The second function
uses the new/recommended method of setting cid name and number, but it has NO
EFFECT. (i.e. the name and number remain at the asterisk default).
Why is this one not working?
Thanks,
MD
==
// Test #1
function setid1($name,$number) {
$newid
= \ . trim( substr( trim( $name ), 0, 15 )
)
. \ .
trim( substr( str_replace( , , $number ), 0, 24 )
)
.;
obj-set_callerid( $newid );
}
// Test
#2
function
setid1($name,$number) {
$obj-cmd(Set(CALLERID(name)=\$name\);
$obj-cmd(Set(CALLERID(num)=\$number\);
}
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Re: [asterisk-users] Help with semaphores
Mitch Thompson wrote:
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as load
generators and call sinks when testing equipment. However, the
equipment we are testing gets more and more complex, and the scripted
scenarios the Ameritecs give have become a limiting factor for testing.
Therefore, Asterisk was chosen as a possible solution (we're a cheap lab).
I've been learning Asterisk as I go, but I've learned a lot. Here's the
basic scenario:
We are using an Asterisk (AAH 2.8, specifically) to sink calls. I do
this by taking the ${EXTEN} and breaking it down by sections until I get
to the last 4 digits (i.e., 2105551212). Once I get to the 4-digit
extension, I am trying to set a flag, or semaphore, to do Busy/Idle
testing. Here is my extensions_custom.conf fragment:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at the
beginning of [from-trunk-custom] to the full dialed digits in ${EXTEN},
before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from the
beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new extension
called Idle, where we do a Random to decide whether to simulate no one
home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then hangup.
The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created in
priority 102 (n+101).
I've tried to implement semaphores by using both local and global
variables, but it doesn't seem to work.
My ultimate question: Is anyone doing something similar, and what did
you do to implement the busy/idle.
I appreciate any help anyone can offer.
Mitch Thompson
Hi,
dont know if this is what you looking for but, there is something called macroexclusive, new in 1.4,
written by Steve Davies.
Read the file in asterisk-1.4.0/docs.
HTH
--
thanks,
Yusuf
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[asterisk-users] asterisk 1.4 and r2mfc or unicall
Hi Guys.. I want to see what the R2mfc community has been up to. Anybody moved to 1.4? what have you done regarding unicall? Any updates or are you stuck with 1.2.X too? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Toll-free dialing via PRI problem
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Irvin Sent: Wednesday, January 31, 2007 10:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Toll-free dialing via PRI problem Sheepishly, that was the magic bullet. Thanks Trevor!! Tim Trevor Peirce [EMAIL PROTECTED] wrote: Jerry Jones wrote: From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. So instead of Dial(Zap/g1/1800xxx,,r) just do Dial(Zap/g1/1800xxx,,) so early audio can make it through. Unless there's more to the puzzle? My setup was also fixed. Thanks Trevor! :-D Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX/Debian Aborts Call While Connecting
The point is that the SIP carrier side gets the abort *before the SIP carrier can complete the connection*. That doesn't take 45s. It takes something like a few seconds. What is causing my (Asterisk) side to abort right after completing registration? On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote: Yeah, your waittime parameter in your call file is set to 45 seconds. db On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote: I used the FreePBX on Debian HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from /var/log/asterisk/full]: [...] -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax from PAP2 through a zap channel to PSTN
On 2/1/07, Chung-lai Chan [EMAIL PROTECTED] wrote: Hello all, Can I send fax from PAP2 through a zap channel to PSTN? I have tried but it is not successful. Thank you for your help! Lai Try to remove echo cancellation (any type of cancellation) and VAD. I got good answer receiving fax as sip client behind a PAP2. -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: why there havn't app_meetme.so fileaboutasterisk1.4.0?
Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma card dying after 1hour
Hi Jon, Did you find any solution for your problem? -- Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schopzinsky Sent: Friday, January 26, 2007 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with semaphores
Mitch Thompson wrote:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from
the beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new
extension called Idle, where we do a Random to decide whether to
simulate no one home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then
hangup. The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created
in priority 102 (n+101).
Here's what I would do...
First, no System calls. Stay within asterisk.
I doubt this will get rid of all race conditions, but I imagine it would
at least reduce them.
exten = 1212,1,GotoIf($[${DB(/tmp/${orig_num})} != ${UNIQUEID}]?abort)
exten = 1212,1,Set(${DB(/tmp/${orig_num})}=${UNIQUEID})
exten = 1212,n,GotoIf($[${DB(/tmp/${orig_num})} != ${UNIQUEID}]?abort)
exten = 1212,n, Goto(Idle,1)
exten = 1212,n(abort),Busy()
Regards,
Trevor
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Re:[asterisk-users] CDR - uniqueid
hi, i think, by default, 'uniqueID' is created by the asterisk. if this is correct, you would (eventually) have non-uniqueID's i saw somewhere in the wiki that someone suggested a change (in the code ?) so that 'uniqueID' would be generated by the database. unique-id being the primary key and autogenerated, it IS unique (cdr+mysql or similar query) it was a rather small change shaoul jacobson -- Initial header --- From : [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com,asterisk-dev@lists.digium.com CC : Date : Thu, 1 Feb 2007 12:15:21 +0100 Subject : [asterisk-users] CDR - uniqueid Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Parcina Lama Computers Split --- Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0?
Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CDR - uniqueid
Hi this was on the mailing list. Some one posted, I didn't tested my self but I believe it should work. I believe that you can set a systemname=blah in asterisk.conf and that will be pre-pended with a dash to the uniqueid. For example: systemname=node1 the uniqueid might look like node1-11673478409.12 This would be 100% unique as long as you gave each system a different name. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jacobso1 Sent: 01 February 2007 14:25 To: asterisk-users Subject: Re:[asterisk-users] CDR - uniqueid hi, i think, by default, 'uniqueID' is created by the asterisk. if this is correct, you would (eventually) have non-uniqueID's i saw somewhere in the wiki that someone suggested a change (in the code ?) so that 'uniqueID' would be generated by the database. unique-id being the primary key and autogenerated, it IS unique (cdr+mysql or similar query) it was a rather small change shaoul jacobson -- Initial header --- From : [EMAIL PROTECTED] To : Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com,asterisk-dev@lists.digium.com CC : Date : Thu, 1 Feb 2007 12:15:21 +0100 Subject : [asterisk-users] CDR - uniqueid Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Parcina Lama Computers Split --- Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros http://www.scarlet.be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] musiconhold restarts for every extension
Benko, You can put multiple files in the MOH directory giving your listener a good chance of getting a new piece of music each time he is on hold. Asterisk picks one of your files randomly. Regards, Brian - Original Message - From: Benko [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, January 31, 2007 6:16 AM Subject: Re: [asterisk-users] musiconhold restarts for every extension On Tue, 30 Jan 2007 12:04:30 -0600 Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: While in 1.2.9 musiconhold was playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: As I understand it, this is the way Native Music on Hold works. mpg123 based MoH does not restart for each call. Well, it was working perfectly with Native MOH in 1.2.9. Judging the two replies, this is a bug(imho). I think it should at least be optional if you want it to be restarted or not(if there's anyone who needs the current behaviour). I don't want to sound like an dissatisfied customer however, i honour that asterisk is mostly voluntary work and since i'm not a programmer i try to contribute with feedback at least. Regards Thx Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange caller display
Rilawich Ango wrote: Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows asterisk when I make a call to the receiver. I wonder why asterisk shows in the display as I haven't set any word - asterisk in any configuration file. How to remove that word from the receive end if it is a default word? Below is the log dump from ngrep. There is no asterisk in the from header except the option message. I wonder why asterisk will be shown in the receiver end's screen. ango callerid on SIP channels defaults to asterisk. The only way to override it is to set it in your sip.conf. Though I think that if you get callerid from the calling device, that would override the default as well. I think the order of precedence is (highest to lowest): sip.conf callerid for calling device calling device's transmitted callerid default - asterisk Earle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText() question
I have an F3000 phone utstarcom and sending a text message to it. All is working but there is a line of sender: asterisk. How do I control what this line says? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3 PCI slot with exclusive IRQ ? please advice!
Hi, I'm looking for an hardware platform for an * installation that should have at least 3 PCI slot with no irq sharing whatsoever. Hardware raid 1 with hot swap is a premium, but not mandatory ... What would you choose? compaq/hp ? Dell ? Ibm ? Tnx for any advice on this matter! -- I migliori saluti, Scrivi a: Alessio [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SendText() question
From: Jerry Geis [EMAIL PROTECTED] I have an F3000 phone utstarcom and sending a text message to it. All is working but there is a line of sender: asterisk. How do I control what this line says? Try sip.conf, callerid=... Yuan Liu THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging to /dev/ttyS0
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can pick up CallerID. How can I redirect the log output of asterisk to /dev/ttyS0 or /dev/console? Many thanks, Neil safeharbour IT Ltd Your IT Department fax: 0845 867 2891 mob: 07812 114784 voip: [EMAIL PROTECTED] email: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] web: http://www.safeharbourit.co.uk/ www.safeharbourit.co.uk The information in this e-mail is confidential and may be legally privileged. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients, any opinions or advice contained in this e-mail are subject to the terms and conditions expressed in any applicable governing terms of business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SendText() question
It needed BOTH the text callerid and numerid callerid to display the text form. Callerid: Some name number At first I was only supplying the name. Works fine. Thanks, Jerry /From: Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users // //I have an F3000 phone utstarcom and sending a text message to it. //All is working but there is a line of sender: asterisk. //How do I control what this line says? / Try sip.conf, callerid=... Yuan Liu /THanks, // //Jerry/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on G.729 (was: H.264 *Not Patented*)
Hi, I asked some questions here about G.729 earlier in the week, and it looks like it would fit the bill for compressing audio between my * server in colocation and sip phone at home. This is what I want my setup to look like. (Wont make sense unless you are using a fixed width font) [my phone] [asterisk] [third parties] Snom 360-- v 1.4 - ??? SIP IAX/SIP G.729 Don't care (probably something other than G.729, my preferred supplier today likes ulaw and alaw) My phone sees the * box over a relatively slow consumer connectivity link. The * box is colocated and has excellent connectivity. Therefore the tighter compression between * and my phone is important, hence why I want to use g.729 here. The config for my phone, and my preferred voice supplier looks like this : [[[from sip.conf]]] [andydesk] type=friend context=default secret=xxx host=dynamic dtmfmode=rfc2833 username=andydesk mailbox=1001 vmexten=500 disallow=all allow=g729 allow=alaw allow=ulaw allow=gsm regexten=1001 allowreinvite=no [[[from iax.conf]]] [thing] type=friend host=dynamic username=thing secret=xxx trunking=off bridging=on context=thing disallow=all allow=ulaw allow=alaw allow=gsm When I place a call, the other party's line rings as normal. When the other party answers, I get a sip 'denied' packet, and the call is aborted. Asterisk says : No path to translate from SIP/mydeskphone to IAX/myprovider and Had to drop call because I couldn't make SIP/ mydeskphone ompatible with IAX/myprovider. This looks similar to this bug : http://bugs.digium.com/view.php?id=8781nbn=4 What I would expect to happen, is that Asterisk would transcode between the ulaw/alaw party, and me, wanting to listen via g729. Is this what *should* happen ? Worth noting that my provider does not support G.729. Is what is happening a bug ? Any patches I can try to see if they work ? Or is it my config which is broken ? Inbound calls work ok, I guess this is because they are presented as alaw and asterisk is just passing them through (which of course isn't what i really want). Thanks for any suggestions, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi
Any ideas? It should be simple... Cosmin Prund wrote: Hello everyone: using chan_capi 0.7 and asterisk 1.4 Quick question: How can I detect the number of voice channels (B channels) in use at a given time. I'd like to call Busy if two B channels are used on my BRI to give the calling customer a Busy signal. Long question: On my single-line BRI (two channels) I'd like to give the 3rd simultaneous incoming call an busy signal. I already tested and the Busy function works very well (I've set up one of my MSN's to immediately call Busy). I also tested and I'm 100% sure the 3rd call makes it into the box while the other 2 channels are talking, so this is not a Telco problem and it can be fixed locally. Doing this on my side of the line (as opposed to having the Telco issue the Busy signal on my behalf) has an number of benefits: (a) I don't need to talk to the Telco (b) I *know* who called and I can call them back and (c) In a distant future I might use the capi channel's ability to transfer the call to a different POTS line since this doesn't use the B channel. Thanks, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan logic based on caller ID
Hello! Is there any easy way to use the caller ID display info (CALLERID(name) in Asterisk) in dialplan just as we could use the number in: exten = _X./67803287, 1, action I have a SIP GSM device, and when a call comes in, it passes me the caller ID like so: -- Sip message Header: From: 67803287 sip:[EMAIL PROTECTED];tag=... -- Asterisk variables: CALLERID(num) = gsm CALLERID(name) = 67803287 I would like to make a logic based on the caller id, that would also work in the case of zap devices that set variables like: CALLERID(num) = 67803287 CALLERID(name) = Anyone has a clue (without having to complicate things too much)? Thank you, François. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap Load/Stress scripts?
Are there any scripts out there that would help me stress test two boxes that are setup back to back with 4 PRI connections? We're having problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm tired of testing them in a production environment. As Sangoma provides firmware updates (and various other shots in the dark) I'd like to be able see if the problem is fixed in an isolated environment. I just need a way to simulate call volume on 4 t1s. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get the status of failed call files
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial()
isn't
used) and channel_status doesn't seem to be any good.
thanks in advance.
the event you received for OriginateFailure has a 'Reason: ' code.
that code breaks down as
0 = UNKNOWN FAILURE or DISCONNECT
3 = AST_CONTROL_RINGING (no answer)
5 = AST_CONTROL_BUSY
1 = AST_CONTROL_HANGUP
8 = AST_CONTROL_CONGESTION
i didn't originate the call with manager, but with a call file, so i
can't
get manager events. i there any other way of finding this out?
if DIALSTATUS or HANGUPCAUSE don't have what you need then you will need
to mod pbx.c
look for a section of code that looks like this
/* create a fake channel and execute the
failed extension (if it exists) within the requested cont
/* check if failed exists */
if (ast_exists_extension(chan, context,
failed, 1, NULL)) {
chan = ast_channel_alloc(0);
if (chan) {
ast_copy_string(chan-name,
OutgoingSpoolFailed, sizeof(chan-name));
if (!ast_strlen_zero(context))
ast_copy_string(chan-context, context, sizeof(chan-context));
ast_copy_string(chan-exten,
failed, sizeof(chan-exten));
chan-priority = 1;
ast_set_variables(chan, vars);
insert pbx_builtin_var here --
ast_pbx_run(chan);
since DIALSTATUS and HANGUPCAUSE are both protected, you will probably
have to create another such as FAILEDCODE.
i hope this helps.
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Re: [asterisk-users] Zap Load/Stress scripts?
Use auto dial. You can have as many calls as you wish. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Porier, Jeremy M. wrote: Are there any scripts out there that would help me stress test two boxes that are setup back to back with 4 PRI connections? We're having problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm tired of testing them in a production environment. As Sangoma provides firmware updates (and various other shots in the dark) I'd like to be able see if the problem is fixed in an isolated environment. I just need a way to simulate call volume on 4 t1s. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Dial Plan
On 31 Jan 2007, at 14:32, Rob Schall wrote: Perfect. Here's a quick and hopefully doable followup question. We have Polycom Soundpoint 501 phones. Is there a way to have a phone check 2 voicemail boxes? If we have a queue, and we want the MWI to show for say that users's extension 1000 and the special billing vm box of 2000. In the absence of any other replies, I'll mention that I'm pretty sure that the Snoms will check multiple voicemail-boxes for the MWI if you comma-seperate the mailbox= options in sip.conf e.g. [EMAIL PROTECTED],[EMAIL PROTECTED] Give that a try ! Cheers -a -- Regards, Andy Davidson http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Jerry Geis [EMAIL PROTECTED] It needed BOTH the text callerid and numerid callerid to display the text form. Callerid: Some name number Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Yuan Liu At first I was only supplying the name. Works fine. Thanks, Jerry /From: Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users // //I have an F3000 phone utstarcom and sending a text message to it. //All is working but there is a line of sender: asterisk. //How do I control what this line says? / Try sip.conf, callerid=... Yuan Liu /THanks, // //Jerry/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Give Busy to the 3rd call on a BRI using chan_capi
On Thu, 1 Feb 2007, Cosmin Prund wrote: Any ideas? It should be simple... It is easy: read the README in chan-capi.org package ;-) Just look into the variable BCHANNELINFO and you will know if it is a call without b-channel (the third call). Armin Cosmin Prund wrote: Hello everyone: using chan_capi 0.7 and asterisk 1.4 Quick question: How can I detect the number of voice channels (B channels) in use at a given time. I'd like to call Busy if two B channels are used on my BRI to give the calling customer a Busy signal. Long question: On my single-line BRI (two channels) I'd like to give the 3rd simultaneous incoming call an busy signal. I already tested and the Busy function works very well (I've set up one of my MSN's to immediately call Busy). I also tested and I'm 100% sure the 3rd call makes it into the box while the other 2 channels are talking, so this is not a Telco problem and it can be fixed locally. Doing this on my side of the line (as opposed to having the Telco issue the Busy signal on my behalf) has an number of benefits: (a) I don't need to talk to the Telco (b) I *know* who called and I can call them back and (c) In a distant future I might use the capi channel's ability to transfer the call to a different POTS line since this doesn't use the B channel. Thanks, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get the status of failed call files
Richard Lyman wrote:
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial()
isn't
used) and channel_status doesn't seem to be any good.
thanks in advance.
the event you received for OriginateFailure has a 'Reason: ' code.
that code breaks down as
0 = UNKNOWN FAILURE or DISCONNECT
3 = AST_CONTROL_RINGING (no answer)
5 = AST_CONTROL_BUSY
1 = AST_CONTROL_HANGUP
8 = AST_CONTROL_CONGESTION
i didn't originate the call with manager, but with a call file, so i
can't
get manager events. i there any other way of finding this out?
if DIALSTATUS or HANGUPCAUSE don't have what you need then you will need
to mod pbx.c
look for a section of code that looks like this
/* create a fake channel and execute the failed
extension (if it exists) within the requested cont
/* check if failed exists */
if (ast_exists_extension(chan, context, failed,
1, NULL)) {
chan = ast_channel_alloc(0);
if (chan) {
ast_copy_string(chan-name,
OutgoingSpoolFailed, sizeof(chan-name));
if (!ast_strlen_zero(context))
ast_copy_string(chan-context,
context, sizeof(chan-context));
ast_copy_string(chan-exten, failed,
sizeof(chan-exten));
chan-priority = 1;
ast_set_variables(chan, vars);
insert pbx_builtin_var here --
ast_pbx_run(chan);
since DIALSTATUS and HANGUPCAUSE are both protected, you will probably
have to create another such as FAILEDCODE.
i hope this helps.
ok. thanks. HANGUPCAUSE only seems to return 0. if it's not possible to
find this out whilst using call files (as it appears) i'll take a look
at using manager.
--
- Rich Doughty
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Re: [asterisk-users] how to get the status of failed call files
*snipped ast_set_variables(chan, vars); insert pbx_builtin_var here -- ast_pbx_run(chan); since DIALSTATUS and HANGUPCAUSE are both protected, you will probably have to create another such as FAILEDCODE. i hope this helps. oops, almost forgot. and you need to fill it with 'reason'. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Load/Stress scripts?
take a look on Originate command for Asterisk manager interface to get web page generating calls between the two boxes. Easier I believe is to use SIPp to be used as an UAC that starts dialing to your box1 and in the dialplan of this box make a dial for a Zap channel on Box2. You need to compile sipp with media streaming and authentication or if you just want first to test you may provide an extension named service in the context defined in general section of your sip conf for external calls coming to your asterisk server without authentication: http://sipp.sourceforge.net/doc/reference.html#Installing+SIPp - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand without authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay - *With PCAP playhttp://sipp.sourceforge.net/doc/reference.html#pcapplayand authenticationhttp://sipp.sourceforge.net/doc/reference.html#authenticationsupport *: # gunzip sipp-xxx.tar.gz # tar -xvf sipp-xxx.tar # cd sipp # make pcapplay_ossl Example: - Sipp being used as a SIP user agent Client: - Call Duration 1ms - Dialing Calls with RTP using ulaw ./sipp -sf uac_pcap.xml -d 1 192.168.34.6 -trace_err Where this IP is my * . On 2/1/07, Mik Cheez [EMAIL PROTECTED] wrote: Use auto dial. You can have as many calls as you wish. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Porier, Jeremy M. wrote: Are there any scripts out there that would help me stress test two boxes that are setup back to back with 4 PRI connections? We're having problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm tired of testing them in a production environment. As Sangoma provides firmware updates (and various other shots in the dark) I'd like to be able see if the problem is fixed in an isolated environment. I just need a way to simulate call volume on 4 t1s. Thanks, Jeremy Porier Senior Director of IST Colorado Christian University [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on G.729 (was: H.264 *Not Patented*)
On 2/1/07, Andy Davidson [EMAIL PROTECTED] wrote: What I would expect to happen, is that Asterisk would transcode between the ulaw/alaw party, and me, wanting to listen via g729. Is this what *should* happen ? Worth noting that my provider does not support G.729. Is what is happening a bug ? Any patches I can try to see if they work ? Or is it my config which is broken ? How many g729 licenses do you have? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dell Servers
Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse [EMAIL PROTECTED] wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging to /dev/ttyS0
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
I think you might be better off with a System() call in your dial plan such as:
System(echo ${CALLERIDNUM} /dev/ttyS0)
That will send the callerID number followed by a new line. You can of
course change the format to your desire. Make sure /dev/ttyS0 is
writable by the asterisk user, and is also properly set up (baud rate,
bits, ...).
--Luki
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[asterisk-users] Dial option G - Passing parameters?
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers (Customer)
Customer identifies himself, and now I use Dial w/ the G option:
Dial(Zap/g9/${agentext}|60|mG(Agent_Xfer^s^1)
Customer hears MOH while the Dial app gets the agent on the line
My destination context looks like this:
[Agent_Xfer]
exten = s,1(Customer),Meetme({$ConfRoom}|qM)
exten = s,2(Agent),Macro(Connect_to_agent,${Customerid},${ConfRoom})
Customerid and ConfRoom are channel variables that are set in the
Originate action and at the start of the dialplan processing,
respectively.
The idea is to put the customer in a conference room, listening to MOH,
until I can get an agent on the line. (This part works pretty well.)
The agent is an extension on a legacy PBX, so a simple Dial with a macro
has undesired side effects. (Specifically, the customer hears ringing
or the legacy PBX's MOH, depending upon the status of the transfer.)
Putting the customer in a conf room, listening to music, is the best
solution I can think of.
The problem is that I don't know how to get the two channel variables
over to the Agent leg of the call. I don't see anything in the docs
about the G option accepting arguments to pass to the called leg. Is
there any way that I can get the two variables' values over to the
called leg?
-MC
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Re: [asterisk-users] Dell Servers
On Thu, 1 Feb 2007, Eric Rousse wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? I bought a Dell 2850 as a pbx server and it just sucks IMHO The stupid thing has only 3 pci slots and even with only 3 pci slots Dell managed to have a shared irq on every slot, 1 for the scsi controller and one for each nic The result of this 'nice' piece of work is dreadfull irq hit/miss results in zttest, it barely meets the minimum requirement and i do get complaints of dropped calls on my pri I need to pass some options to the kernel at boot time to improve things, without extra options the results from zttest were unacceptable My spare pbx is a lowly Athlon XP 2600 with an Asus A7V8X-X mobo in it and it's scores with zttest are considerably better (but not full 100% hits) I know that everybody on the list will now start recommending me to buy Sangoma hardware but firstly I hate compiling extra modules and it doesn't make it right that the Dell hardware just sucks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
Hello, Well we're planning either use some PRI lines or IP Trunk, where not sure yet. For the PRI lines we will probably use a Wildcard TE412P, so PCIe For the IP Trunk, not sure yet I don't have a lot of info in that regards. I'm also planning to put an extra server with some cards to connect to a SAN with Fiber channel, not decided yet between Fiber channel and Gigabit switch dedicated. Anyway... Matt Florell a écrit : Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse [EMAIL PROTECTED] wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Talkoff
Hello all, I'm trying to see if I can finally get rid of a talkoff problem that I've been having with my Asterisk server since I started messing with it over a year ago. Currently, I'm running it on SuSE 10.1 box with Asterisk 1.4. I'm using Snom 360s with the set. My setup is one where the PSTN connects to a legacy PBX (Definity), then connects, via a T1 cable, to a 4 port T1 card in the Asterisk. Doing research over the past few months, I've seen that I should use dtmfmode=rfc2833 and ensure that I use ulaw. The sip.conf section referring to my extension looks like this: [5257] type=friend secret=5257 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=5257 host=dynamic dtmfmode=rfc2833 disallow=gsm,alaw dial=SIP/5257 context=from-internal canreinvite=no callerid=device 5257 My Snom 360 is set to use G.711u first, which I presume is ulaw. I don't notice the problem on every call, just some calls. It always seems to be to calls going outside, but that's most of the calls I deal with( as opposed to staying on the Asterisk or bridging to the Definity). The person calling me never hears the tones, just me. What can I look at to get more clues? Cheers, Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
We have a 2850 in a productive environment with a BNE1 performing well (OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu Edgy). You only have to blacklist some hotplug kernel modules and yes, we do have very long pings (1 ping per week with a check rate of 10min per SNMP). But that does happen very rare and I never noticed any dropped calls or bad audio quality. The 2850 is running on SCSI, the 2950 on an SAS RAID. In general I like the Dell machines, also with asterisk on them. The only thing is that Openmanage ist quite bad to install but that's nothing asterisk specific but linux related. Does that help? best regards, Christophorus Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? I bought a Dell 2850 as a pbx server and it just sucks IMHO The stupid thing has only 3 pci slots and even with only 3 pci slots Dell managed to have a shared irq on every slot, 1 for the scsi controller and one for each nic The result of this 'nice' piece of work is dreadfull irq hit/miss results in zttest, it barely meets the minimum requirement and i do get complaints of dropped calls on my pri I need to pass some options to the kernel at boot time to improve things, without extra options the results from zttest were unacceptable My spare pbx is a lowly Athlon XP 2600 with an Asus A7V8X-X mobo in it and it's scores with zttest are considerably better (but not full 100% hits) I know that everybody on the list will now start recommending me to buy Sangoma hardware but firstly I hate compiling extra modules and it doesn't make it right that the Dell hardware just sucks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Talkoff
On Thursday 01 February 2007 4:01 pm, McGhee, Stefano wrote: I'm trying to see if I can finally get rid of a talkoff problem that I've been having with my Asterisk server since I started messing with it over a year ago. Currently, I'm running it on SuSE 10.1 box with Asterisk 1.4. I'm using Snom 360s with the set. My setup is one where the PSTN connects to a legacy PBX (Definity), then connects, via a T1 cable, to a 4 port T1 card in the Asterisk. What is the manufacturer and model of the 4-port T1 card? I have had talkoff with the TE406 (1st gen echo canceller), and have heard of talkoff occurring with relaxdtmf=yes in zapata.conf. I have *not* had talkoff issues at all with Sangoma's A104d nor with Digium's TE407 cards. Digium's TE406 most certainly had issues though. So: - if you have a TE406, disable hardware DTMF detection - check that you do NOT have relaxdtmf=yes in zapata.conf -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dell Servers
We have the 2950. It came with only 2pcix ports. And if you need to power an fxs card, you need to route wires around. It wasn't easy to work with. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Thursday, February 01, 2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dell Servers Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse [EMAIL PROTECTED] wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
We built some systems based on Dell 2950's and they ran fine. We put a TE110P in a Dell 860 last week, and it makes a noise on the outbound part of the call. Not on the inbound, which is really odd. The 860 works perfectly with a TE210P in it though. (This fault has been logged with Digium.) later, PaulH On Thu, 2007-02-01 at 14:57 -0500, Eric Rousse wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Talkoff
What is the manufacturer and model of the 4-port T1 card? I have had talkoff with the TE406 (1st gen echo canceller), and have heard of talkoff occurring with relaxdtmf=yes in zapata.conf. Hey there. I do believe it it a Digium TE406 with Echo Canceller. I can't remember how many times I've read about relaxdtmf :-) Yes, that's set to no in zapata.conf. So: - if you have a TE406, disable hardware DTMF detection Interesting about the TE406 though. How does one turn off the hardware DTMF detection? I imagine it's in the driver/module config somewhere. Thanks, Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server hardware choice,
Any mid-level server (kinda 3ghz 2GB ram) you have been wonderfully happy with? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please help parse this GotoIf line
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings depending on the
caller id. I have one set up and working for 'Internal' calls but
unfortunately the same tone will ring if caller id is absent on a call.
My solution is to insert a caller id number of 'NOCID' if there is no
caller id to have separate ring tones for 'NOCID' and Internal' calls.
I have gotten this far for the nth line in my extensions.conf
[telasip-in] context but need help with the syntax.
In Asteriskish it would look something like:
exten = s,n,GotoIf( NO ${CALLERID} then SetCIDNum(NOCID)
I really wish to be able to pick up an Internal call without thought but
don't really like getting NOCID sales and other annoying calls.
Note: I looked at privacymanager and will try it if the above can't be
made to work.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Using Local Channels with Originate
I have been trying to get a DIALSTATUS output from a call started with
originate. I searched a fair bit and have found several references to using
local channels to do this. However, I could not find enough of the specifics
to get it working myself.
What I need to do is dial a zap channel and run various scripts if the
channel is answered, busy, no-answer,etc.
Here is the dial plan I am using:
[outdialer]
exten = 100,1,Dial(ZAP/4/1234567)
exten = 100,n,DeadAgi(notdeadyet.py)
exten = 100,n,Hangup
[dialerplan]
exten = s,1,AGI(showstatus.py|${DIALSTATUS})
exten = s,n,Hangup
Here are the manager commands I am using:
Action: login
Username: test
Secret: nottelling
Action: originate
Channel: Local/[EMAIL PROTECTED]/n
Context: dialerplan
Extension: s
Priority: 1
Action: logoff
The notdeadyet.py script never runs. The ${DIALSTATUS} passed into
showstatus.py is empty. I don't understand what I did wrong.
Thanks in advance for your help. I am stumped by this.'
-Brian
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Re: [asterisk-users] Please help parse this GotoIf line
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings depending on the
caller id. I have one set up and working for 'Internal' calls but
unfortunately the same tone will ring if caller id is absent on a call.
My solution is to insert a caller id number of 'NOCID' if there is no
caller id to have separate ring tones for 'NOCID' and Internal' calls.
I have gotten this far for the nth line in my extensions.conf
[telasip-in] context but need help with the syntax.
In Asteriskish it would look something like:
exten = s,n,GotoIf( NO ${CALLERID} then SetCIDNum(NOCID)
I think something like
exten = s,n,GotoIf($[${CALLERID} = ]?anonymous:withnumber)
exten = s,n(anonymous),Set(CALLERID(num)=NOCID)
exten = s,n(withnumber),..
should do the trick.
BR
Anselm
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Re: [asterisk-users] 3 PCI slot with exclusive IRQ ? please advice!
Alessio Focardi wrote: Hi, I'm looking for an hardware platform for an * installation that should have at least 3 PCI slot with no irq sharing whatsoever. Use an industrial PC with a backplane bus. You can easily get 3-4 usable slots in a 2U and 10-14 slots if you use a 4U. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with semaphores
Trevor Peirce wrote:
Mitch Thompson wrote:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this number, return busy
exten = 1212,102,System(echo ${UNIQUEID} /tmp/${orig_num}) ;
basically, create a file in /tmp whose name is the full number from
the beginning. In this case, the full
; filename would be /tmp/2105551212. I don't really care about the
contents, though.
exten = 1212,103, Goto(Idle,1) ; from here, we jump to a new
extension called Idle, where we do a Random to decide whether to
simulate no one home (ring no answer) or
; we send ring for
about 10 seconds, then Answer() and play some .wav files, then
hangup. The last thing we do in either case is to delete
; the
/tmp/${orig_num} file.
The above code works very well at low call volumes. However, I'm
running into race conditions at high call volumes where several calls
are getting through the test in priority 1 before the file is created
in priority 102 (n+101).
Here's what I would do...
First, no System calls. Stay within asterisk.
I doubt this will get rid of all race conditions, but I imagine it
would at least reduce them.
exten = 1212,1,GotoIf($[${DB(/tmp/${orig_num})} !=
${UNIQUEID}]?abort)
exten = 1212,1,Set(${DB(/tmp/${orig_num})}=${UNIQUEID})
exten = 1212,n,GotoIf($[${DB(/tmp/${orig_num})} !=
${UNIQUEID}]?abort)
exten = 1212,n, Goto(Idle,1)
exten = 1212,n(abort),Busy()
Regards,
Trevor
Trevor,
Thank you very much for your response. Yes, I was worried about the
performance hit of the System() call, but as I said, I tried doing it
with Global and/or local variables, and just wound up confusing myself.
I'm not a programmer by trade or nature, so I knew up front it was
dirty code. Your method of staying with Asterisk looks like something
I can easily switch to in one exchange context and give it a quick try.
Thanks again!
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[asterisk-users] Asterisk Scaling/Load Balancing for iax soft clients
Hi There, I am interested to know the solutions available for a large call centre with more than 200 seats running asterisk and iax soft clients(eg .idefisk). All the calls are through soft clients , so there is no PSTN requirement/connectivity.This is a pure iax implementation. 1. How to scale a single queue across multiple asterisk boxes at the same time keeping track of all the agents 2. To load balance iax soft clients do we have something similar to OpenSER Any pointers will be highly appreciated. Thanks in advance. Cheers, boneyM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed.
hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC] astman.c - astman.o [CC] md5.c - md5.o [LD] astman.o md5.o - astman [CC] smsq.c - smsq.o [CC] strcompat.c - strcompat.o [LD] smsq.o strcompat.o - smsq [CC] stereorize.c - stereorize.o [CC] frame.c - frame.o [LD] stereorize.o frame.o - stereorize [CC] streamplayer.c - streamplayer.o [LD] streamplayer.o - streamplayer make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed. make: *** [utils] Aborted ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] API Originate Action - distinguishing between No Answer and Invalid phone number
I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no answer and an invalid? For me, a 'failed' attempt is dialing an invalid number, and I'd like the CDRs to reflect that. I'd like a no answer to be as 'successful' as a busy. I'm familiar with the 'OriginateFailure' event and it's 'Reason' field, but I don't know how to get that reason into the CDR. Is that possible? Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0?
Bill Gibbs,hello Thank you so much. According to this method , I get the app_meetme.so . === 2007-02-01 22:49:43 您在来信中写道：=== Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-02 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI script callerid question
Michelle Dupuis wrote: I am trying to set callerid from a PHP script, using one of two functions as shown below (setid1 and setid2). The first function works great with regular names and numbers, BUT, if I call the function with (Test,UnknownNumber), the cid number gets set to asterisk. Why is my passed number parameter not being accepted in this case? Try to call the function with (Test,) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API Originate Action - distinguishing between No Answer and Invalid phone number
On 2/1/07, Michael Collins [EMAIL PROTECTED] wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy.
The ${DIALSTATUS} channel variable stores the result of the dial attempt:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
You can store it on the CDR's userfield column using the cdr function:
Set(CDR(userfield)=${DIALSTATUS})
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Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] windows SIP Softphones ?
hi all what do must win32 clients use as a free or OSS sip softphone ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] windows SIP Softphones ?
Dennis Kavadas wrote: hi all what do must win32 clients use as a free or OSS sip softphone ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Usually have good luck with xlite http://xten.com/index.php?menu=download signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] windows SIP Softphones ?
Hi Try that one http://www.counterpath.com/index.php?menu=Productssmenu=xlite Bruno C. Branco -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Kavadas Sent: February 02, 2007 10:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] windows SIP Softphones ? hi all what do must win32 clients use as a free or OSS sip softphone ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX/Debian Aborts Call While Connecting
On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote: The point is that the SIP carrier side gets the abort *before the SIP carrier can complete the connection*. That doesn't take 45s. It takes something like a few seconds. What is causing my (Asterisk) side to abort right after completing registration? On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote: Yeah, your waittime parameter in your call file is set to 45 seconds. db On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote: I used the FreePBX on Debian HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from /var/log/asterisk/full]: [...] Alright, take a look the **Lines: **Line 1: Your dial sequence clearly shows the 45sec timeout value being applied as the second value in the dial plan SIP/[EMAIL PROTECTED]|45| -- Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/[EMAIL PROTECTED]|45| M(say-call-2-digits^17182335097)g) in new stack **Line 2: The timer has expired 45000ms is the same 45 second timer that was set for timeout Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in 45000 ms Line 3: The call is dropped towards the carrier. Maybe I am missing something here but it seems you are using a macro with some global variable set for a 45 second wait time for outbound calls. Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] API Originate Action - distinguishing between NoAnswer and Invalid phone number
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork
Sent: Thursday, February 01, 2007 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] API Originate Action - distinguishing between
NoAnswer and Invalid phone number
On 2/1/07, Michael Collins [EMAIL PROTECTED] wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy.
The ${DIALSTATUS} channel variable stores the result of the dial attempt:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
You can store it on the CDR's userfield column using the cdr function:
Set(CDR(userfield)=${DIALSTATUS})
Actually, I can't. The dialplan execution goes straight to the 'failed'
extension. When it does so, the DIALSTATUS variable gets cleared out. I have
this in my dialplan:
exten = failed,n,Noop(Dial status is '${DIALSTATUS}')
The log yields this:
-- Executing NoOp(OutgoingSpoolFailed, Dial status is ) in new stack
Is there perhaps a way to make DIALSTATUS persist or get populated when the
dialplan hits the failed extension?
-MC
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Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: Related to callerid: I can't get text ID to work in an analog phone on FXS. I tried the above format, it simply displays the entire string in both numeric and text field (i.e., displays the same string twice). Tried a few other ways, got varied results (some resulting in Unknown). Nothing can get the analog phone to display name in text field and number in numeric field. I'm using TDM400, phone is 27935GE3-B, Zaptel 1.2.10, Asterisk 1.2.12. On a normal line, the phone displays name on one line and number on another. Anyone sending caller ID to FXS? Works fine with my GE29393GE2-A. I think you need the right syntax, in your .conf it should look like callerid=John Doe 1234 Note the quotes around the name. Leo Ain't working. 27935GE3-B simply says unknown or displays a blank if the string contains quote. I know that I can configure a softphone (e.g., Xten) to display correctly, because it has a user id and a display name. Anything similar in Asterisk? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] make: expand.c:489: allocated_variable_append:Assertion `cu
From: Dennis Kavadas [EMAIL PROTECTED] hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? Likely your make version. See this thread http://forums.digium.com/viewtopic.php?t=12707 in forum. Yuan Liu make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC] astman.c - astman.o [CC] md5.c - md5.o [LD] astman.o md5.o - astman [CC] smsq.c - smsq.o [CC] strcompat.c - strcompat.o [LD] smsq.o strcompat.o - smsq [CC] stereorize.c - stereorize.o [CC] frame.c - frame.o [LD] stereorize.o frame.o - stereorize [CC] streamplayer.c - streamplayer.o [LD] streamplayer.o - streamplayer make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed. make: *** [utils] Aborted ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: CallerID to FXS (RE: [asterisk-users] SendText() question)
Leo Ann Boon wrote: Eric ManxPower Wieling wrote: You should not have quotes in Caller*ID info. MOST devices will just ignore the quotes, but a few will refuse to accept Caller*ID with quotes in it. At least one revision of SIP firmware for Cisco phones does this. Thanks for the heads up. On the other hand, there are devices that will treat everything as the number if you omit the quotes. So you'll get gibberish on the phone. I've never seen one. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Clone Asterisk
I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Voipjet ...
I have found that if you don't have the minimum balance required for the voipjet premium server, you get the circuits busy message, you might want to check your balance. On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Please help parse this GotoIf line
Thanks for your reply Anselm. I'll play it with tomorrow.
Let me ask a related question. I also have to assign a calleridnum
(number) of 'Internal' to each extension dialed on an internal to
internal call. They would all have 4 digit calleridnum in the range 4??
( or _4xx in dial plan form ) to be changed to 'Internal'. I'd like to
avoid many lines of code so is there any way to do that with a wild card
or dial plan type?
Larry
Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings depending on the
caller id. I have one set up and working for 'Internal' calls but
unfortunately the same tone will ring if caller id is absent on a call.
My solution is to insert a caller id number of 'NOCID' if there is no
caller id to have separate ring tones for 'NOCID' and Internal' calls.
I have gotten this far for the nth line in my extensions.conf
[telasip-in] context but need help with the syntax.
In Asteriskish it would look something like:
exten = s,n,GotoIf( NO ${CALLERID} then SetCIDNum(NOCID)
I think something like
exten = s,n,GotoIf($[${CALLERID} = ]?anonymous:withnumber)
exten = s,n(anonymous),Set(CALLERID(num)=NOCID)
exten = s,n(withnumber),..
should do the trick.
BR
Anselm
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--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Asterisk cann't redirect the calling party to anothere Exten.
Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL PROTECTED] Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: SIP/voip_out_22-809c Exten: 111 Context: meetme-test Priority: 1 Then , the channel named SIP/voip_out_22-809c has been transfered to the conference 111. But, the channel named SIP/612-5456 has been hangup automatic. The context meetme-test is : [meetme-test] exten = 111,1,Answer exten = 111,n,MeetMe(111,pdMX) exten = 111,n,Hangup I want to redirect both channels to the conference 111. What's wrong it? With Regards, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to Clone Asterisk
Assuming some defaults... your results may vary. /etc/asterisk = Configs /var/spool/asterisk = Voicemail, other spool files /var/lib/asterisk = Licenses (G729 for example), stock sounds, astdb, etc From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert DeVries Sent: Thursday, February 01, 2007 21:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Clone Asterisk I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging to /dev/ttyS0
On Thu, Feb 01, 2007 at 03:32:22PM -, Neil Tancock wrote: Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can pick up CallerID. How can I redirect the log output of asterisk to /dev/ttyS0 or /dev/console? If you can't simply put /dev/ttyS0 or /dev/console as a log file in logger.conf (make sure that no harm is done on log rotation), log things to syslog and from syslog to those devices. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
