Re: [asterisk-users] OT - In which countries are ISDN subaddressesused ?

2009-09-28 Thread Geoff Lane
On Tuesday, September 29, 2009, C F wrote: > You say no reliable internet, if you can get ISDN wouldn't the > providers offer Internet thru the same ISDN Physical links? Some countries (e.g. UK) don't offer DSL or ATM over ISDN physical links. If a physical circuit is used for ISDN, the only way

Re: [asterisk-users] OT - In which countries are ISDN subaddressesused ?

2009-09-28 Thread C F
On Mon, Sep 28, 2009 at 11:51 PM, Alec Davis wrote: > I'm interested, and I expect others will be on how you might use it. > > Our use on the mantis bug, is to allow 3 ISDN connected sites (no reliable > internet) each running asterisk, to dial other staff members in the other You say no reliabl

Re: [asterisk-users] OT - In which countries are ISDN subaddressesused ?

2009-09-28 Thread Alec Davis
I'm interested, and I expect others will be on how you might use it. Our use on the mantis bug, is to allow 3 ISDN connected sites (no reliable internet) each running asterisk, to dial other staff members in the other branches. The key to this working is the subaddress being populated with the da

Re: [asterisk-users] digium fax: failed to queue document

2009-09-28 Thread sean darcy
On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg wrote: > On Mon, Sep 28, 2009 at 12:30 PM, sean darcy wrote: >> Well one of the problems it that SendFax doesn't like the tiff file(BTW, >> SendFax from app_fax.so gives you clue what the problem is). It requires >> a special sort of tiff file. I n

Re: [asterisk-users] OT - In which countries are ISDN subaddresses used ?

2009-09-28 Thread Olivier
2009/9/26 Alec Davis > When did that happen? Added to libpri, someone beat me to it. > Hello Alec, In this question, I was refering to your ongoing work I'm sorry if this question let anyone believe this ISDN subaddresswas already done and pushed into libpri trunk. Regards > > Alec Davis >

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Barry Miller
On Mon, Sep 28, 2009 at 04:09:43PM -0500, Shaun Ruffell wrote: > On 09/28/2009 01:06 PM, Danny Nicholas wrote: > > Funny. The first thing I always do after a reboot is call in from my > > cell to make sure things work. But last night I rebooted and immediately > > tried dialing out (with a TDM842

Re: [asterisk-users] TE121P Blue Alarm/Recovering

2009-09-28 Thread Lee, John (Sydney)
> Many thanks John of Sydney. My pleasure my fellow Asterisker! > I removed the CRC4 and it worked straight away. Can you recommend a CRC > type at all or would it be best to leave it as nothing? Just leave it blank would do. > -Original Message- > From: asterisk-users-boun...@lists.dig

Re: [asterisk-users] TE121P Blue Alarm/Recovering

2009-09-28 Thread Klaverstyn, David C
Many thanks John of Sydney. I removed the CRC4 and it worked straight away. Can you recommend a CRC type at all or would it be best to leave it as nothing? David of Brisbane. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.c

Re: [asterisk-users] TE121P Blue Alarm/Recovering

2009-09-28 Thread Lee, John (Sydney)
1) I have not seen a blue light (usually red/yellow) before on a Digium card and so don't really know what it means. 2) Try to see if you can see any messages coming up from the Asterisk box itself (not thru putty or other remote console). You should see a steady stream of error messages comin

Re: [asterisk-users] Firefox Plugin for Sip Click2Call

2009-09-28 Thread Paul Hales
http://www.noojee.com.au/Page/NoojeeClick Built for firefox. PaulH Stefan Schmidt wrote: > Hello, > > iam searching for an Firefox plugin which can make an sip Invite and > Redirect after 200 OK, so i dont have to use a softphone, just to > initialise a call by clicking on a number > > i've fo

Re: [asterisk-users] TE121P Blue Alarm/Recovering

2009-09-28 Thread Klaverstyn, David C
I'd appreciate it if someone was able to assist. Running the command: dahdi_hardware: pci::08:08.0 wcte12xp+d161:8000 Wildcard TE121 dahdi_scan: [1] active=yes alarms=REC description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VP

Re: [asterisk-users] Inquiry:Asterisk server remote access

2009-09-28 Thread Ivan Stepaniuk
Jeff LaCoursiere wrote: > It would be really cool if iaxmodem would actually answer an incoming > modem call and pass traffic to something like pppd. For those times when > the pstn link is up, but something is wrong with the 'net connection... > I have a working setup of what you descri

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Shaun Ruffell
On 09/28/2009 01:06 PM, Danny Nicholas wrote: > Funny. The first thing I always do after a reboot is call in from my > cell to make sure things work. But last night I rebooted and immediately > tried dialing out (with a TDM842B) and got: > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable t

Re: [asterisk-users] DAHDI channel congested busy

2009-09-28 Thread Landy Landy
I also found this weird, I thought my equipment was the problem. Good to know about this issue so, Digium takes care of the problem. I'm running: asterisk-1.6.1.5 dahdi-linux-2.2.0.2 libpri-1.4.10.1 ___ -- Bandwidth and Colocation Provided b

[asterisk-users] DAHDI channel congested busy

2009-09-28 Thread Jerry Geis
> > Funny. The first thing I always do after a reboot is call in from my > cell to make sure things work. But last night I rebooted and immediately > tried dialing out (with a TDM842B) and got: > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of > type 'DAHDI' (cause 34

Re: [asterisk-users] How to get "Call-ID" SIP header outside "chan_sip" scope ...

2009-09-28 Thread Mauro Sergio Ferreira Brasil
Hello there! I really hate when this happens, but... It seems channel variable "SIPCALLID" will have the info I need, so the changes will be reduced to propagate the channel to ARA driver method "realtime_var_get". If someone have any additional info, or can indicate some problem on using this

[asterisk-users] How to get "Call-ID" SIP header outside "chan_sip" scope ...

2009-09-28 Thread Mauro Sergio Ferreira Brasil
Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the sourc

Re: [asterisk-users] DAHDI Question/Choppy Sound

2009-09-28 Thread David Backeberg
On Mon, Sep 28, 2009 at 2:47 PM, Andrei Verovski (aka MacGuru) wrote: > Right now during test phase I'm using several XTel clients and 1.6.1 server > all on local network. Pings accross local network are stable 0.19xx ms. I've never heard of XTel, but go looking for settings involving vad, and tr

Re: [asterisk-users] DAHDI Question/Choppy Sound

2009-09-28 Thread Andrei Verovski (aka MacGuru)
Hi Right now during test phase I'm using several XTel clients and 1.6.1 server all on local network. Pings accross local network are stable 0.19xx ms. The same Xten clients connected to external SIP server (not mine) working fine. Choppy means repeating fragments of sound are badly distorted

Re: [asterisk-users] DAHDI Question/Choppy Sound

2009-09-28 Thread David Backeberg
On Mon, Sep 28, 2009 at 1:57 PM, Andrei Verovski (aka MacGuru) wrote: > Sound still choppy. Probably I'm missing something very very simple. There are > 2 Ethernet interfaces on the server, ,I do not have any actual Zaptel > devices, but as I understood, dahdi is still need for timing. Define 'ch

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Danny Nicholas
I've read about this problem a few times with DAHDI. Since it only takes about 5 minutes to remake a release of DAHDI, I would keep a release like 2.1.0.4 on hand to fall back on in case this isn't some minor mistake. DAHDI is still "relatively" new (about 18 months) and this rears it's head every

Re: [asterisk-users] Disable/enable CDR in dialplan

2009-09-28 Thread equis software
Thanks! On Mon, Sep 28, 2009 at 2:28 PM, Paul Dugas wrote: > NoCDR > > On Mon, Sep 28, 2009 at 1:10 PM, equis software > wrote: > > Hi, I need to eneble or disable cdr registration dynamicaly in my > dialplan. > > Is there any way to do this? > > Thanks > > > > > > _

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Barry Miller
On Mon, Sep 28, 2009 at 02:26:29PM +0200, Tzafrir Cohen wrote: > On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote: > > I have a similar problem with DAHDI. If my server gets rebooted, I can't > > make any calls until the a call come in from outside. From there I can > > answer the call

Re: [asterisk-users] DAHDI Question/Choppy Sound

2009-09-28 Thread Andrei Verovski (aka MacGuru)
Hi, Tzafrir! I have installed DAHDI kernel modules and dahdi-tools /etc/init.d./dahdi running $ ls /usr/lib/asterisk/modules/res_timing_* results ==> /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so $ dahdi_test results ==> 99.999% 99.990% 99.994% 99.

Re: [asterisk-users] Disable/enable CDR in dialplan

2009-09-28 Thread Paul Dugas
NoCDR On Mon, Sep 28, 2009 at 1:10 PM, equis software wrote: > Hi, I need to eneble  or disable cdr registration dynamicaly in my dialplan. > Is there any way to do this? > Thanks > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digita

[asterisk-users] strange cisco nat issue

2009-09-28 Thread JR Richardson
Hi All, I'm running into a reoccurring nat issue, phones (Linksys, Polycom or Cisco) behind a cisco router, 2600 series, ios 12.4. Sometimes 2 or more phones will have the same outside port number and will register to Asterisk with the same port number. As far as I can tell this is only occurrin

[asterisk-users] Disable/enable CDR in dialplan

2009-09-28 Thread equis software
Hi, I need to eneble or disable cdr registration dynamicaly in my dialplan. Is there any way to do this? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http

Re: [asterisk-users] digium fax: failed to queue document

2009-09-28 Thread David Backeberg
On Mon, Sep 28, 2009 at 12:30 PM, sean darcy wrote: > Well one of the problems it that SendFax doesn't like the tiff file(BTW, > SendFax from app_fax.so gives you clue what the problem is). It requires > a special sort of tiff file. I never found any way to generate a > compliant file with imagema

Re: [asterisk-users] digium fax: failed to queue document

2009-09-28 Thread sean darcy
sean darcy wrote: > Martin wrote: >> u don't change the ${uniquefile} for the second System/Originate >> >> try to add a string to the ${uniquefile} ... >> >> eg >> >> ${uniquefile}0 >> >> Martin > But I generate another unique file in [fax-tx] just before I try to send > the confirm. > > Here's

[asterisk-users] Asterisk complaning about no such host -- never asked to contact the host it complains about

2009-09-28 Thread Örn Arnarson
Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number)

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Landy Landy
> In your case: is the problem reset by restarting asterisk? > 'dahdi > resstart'? The problem does not reset by restarting asterisk. I've noticed that I can call other sip phones but, when trying to call out, I get the same (Busy/Congested/Not-Available) "congested" messege.

[asterisk-users] Voicemail - remove option to save in different folders

2009-09-28 Thread Mike
I am looking to configure the asterisk voicemail system to stop asking for the folder ("work", "personal", etc) in which to save messages when I do save them. Is there any configuration to do this? Mike ___ -- Bandwidth and Colocation Provi

Re: [asterisk-users] GoTo IF

2009-09-28 Thread Doug Lytle
michel freiha wrote: > Dear Doug, > s-process and s-notexist are contexts or what because I'm getting the > below erroe: I guess need to slow down a bit and read. Yes, they are contexts. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary

Re: [asterisk-users] GoTo IF

2009-09-28 Thread Doug Lytle
michel freiha wrote: > Dear Doug, > s-process and s-notexist are contexts or what because I'm getting the > below erroe: This was just an example of what I've use in my dial plan. You'll need to modify it to fit your dial plan. Doug -- Ben Franklin quote: "Those who would give up Essenti

Re: [asterisk-users] GoTo IF

2009-09-28 Thread michel freiha
Dear Doug, s-process and s-notexist are contexts or what because I'm getting the below erroe: [Sep 28 16:35:07] WARNING[1530]: pbx.c:2470 __ast_pbx_run: Channel 'SIP/voxbone.com-092392e0' sent into invalid extension 's-process' in context 'a2billing4', but no invalid handler regards On Mon, Sep

Re: [asterisk-users] GoTo IF

2009-09-28 Thread Klaus Darilion
extension.ael: if (0!=${MYVARIABLE}) { ... } or test for empty/unset variables, use: ${EXISTS()} or ${ISNULL()} regards klaus michel freiha schrieb: > Hi all, > > I need a goto If statement syntax that check if a variable is not null > then go to dialplan 1 else go to dialplan2 > > Reg

Re: [asterisk-users] MeetMe Hints

2009-09-28 Thread Paul Dugas
I suspect the issue I'm having is more specific to the MeetMe app but that's just a guess. app_meetme.c (in 1.6.1.6) calls ast_devstate_changed(AST_DEVICE_INUSE, "meetme:%s", conf->confno) when the first caller enters a conference and ast_devstate_changed(AST_DEVICE_NOT_INUSE, "meetme:%s",

Re: [asterisk-users] GoTo IF

2009-09-28 Thread Miguel Molina
michel freiha escribió: > Hi all, > > I need a goto If statement syntax that check if a variable is not > null then go to dialplan 1 else go to dialplan2 > > Regards Hi, Use the function EXISTS() inside your GotoIf. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+exists Cheers, --

Re: [asterisk-users] GoTo IF

2009-09-28 Thread Doug Lytle
michel freiha wrote: > Hi all, > > I need a goto If statement syntax that check if a variable is not > null then go to dialplan 1 else go to dialplan2 > exten => s,n,GotoIf($["${conference.room}" != ""]?s-process,1:s-notexist,1) Doug -- Ben Franklin quote: "Those who would give up Essentia

[asterisk-users] GoTo IF

2009-09-28 Thread michel freiha
Hi all, I need a goto If statement syntax that check if a variable is not null then go to dialplan 1 else go to dialplan2 Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizon

Re: [asterisk-users] Problems with Digium TDM400 card

2009-09-28 Thread Danny Nicholas
It is probably as simple as doing a manual "modprobe wctdm". _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Angus Asterisk Sent: Sunday, September 27, 2009 11:18 AM To: asterisk-users@lists.digium.com Subject: [asterisk-use

Re: [asterisk-users] MeetMe Hints

2009-09-28 Thread Danny Nicholas
The routine ast_extension_states in main/pbx.c is set up (in 1.4.26.1) to return these values: switch (devstate) { case AST_DEVICE_ONHOLD: return AST_EXTENSION_ONHOLD; case AST_DEVICE_BUSY: return AST_EXTENSION_BUSY; case AST_DEVICE_UN

[asterisk-users] ding Dialled number down a sip channel to a PBX

2009-09-28 Thread Ishfaq Malik
We have a customer who connects PBX boxes (Avaya etc.) to our asterisk server (1.4.17) as a SIP extension. This customer needs the dialled number sent to the PBX as well as number that the call is originating from so he can set up his own routing from his PBX box. I have tried setting both CALL

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Tzafrir Cohen
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote: > I have a similar problem with DAHDI. If my server gets rebooted, I can't make > any calls until the a call come in from outside. From there I can answer the > call and DAHDI works fine afterwards. In your case: is the problem reset b

Re: [asterisk-users] DAHDI congestion problem

2009-09-28 Thread Landy Landy
I have a similar problem with DAHDI. If my server gets rebooted, I can't make any calls until the a call come in from outside. From there I can answer the call and DAHDI works fine afterwards. --- On Mon, 9/28/09, Tzafrir Cohen wrote: > From: Tzafrir Cohen > Subject: Re: [asterisk-users] DAHD

[asterisk-users] simulate calls for testing sripts

2009-09-28 Thread mickael ropars
Hello, I wrote a AGI script under Asterisk in php, and I want to stress this scripts by simulting call, is there a tool or a way on asterisk to simulate answers to call, and also if possible DTMF? NB: I am not using SIP phone but only analog lines. Regards Mickael __

[asterisk-users] Firefox Plugin for Sip Click2Call

2009-09-28 Thread Stefan Schmidt
Hello, iam searching for an Firefox plugin which can make an sip Invite and Redirect after 200 OK, so i dont have to use a softphone, just to initialise a call by clicking on a number i've found some plugins which only works with a softphone installed on the system but nothing which works good wi

Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

2009-09-28 Thread Tzafrir Cohen
On Mon, Sep 28, 2009 at 08:55:17AM +0200, Loic Didelot wrote: > Hello, > now I can reset and load the firmware, but I do not see the entries > in /proc and the genzaptel command ignores the astribank. The dual PRI > card works fine. > > r...@pbx1:~# zaptel_hardware > usb:002/002 xpp_usb-

[asterisk-users] AGI script

2009-09-28 Thread michel freiha
Dear All, I have a perl script running on my asterisk server...This script is running for all incoming calls...It checks if a user is registered on openSIPS server...Else it return a busy tone... I would like to ask you please about the syntax to call a dialplan defined on extensions.conf for a s

Re: [asterisk-users] Know for how long an agent is talking?

2009-09-28 Thread Lenz Emilitri
If you need this from the moment that the agent connected you can measure the length of the leg that goes with the agent. If you need to measure this from the moment the call was answered, you can measure the length of the 'main' cal of the leg (the one that called the command queue()) If you nee

Re: [asterisk-users] dCAP Exam

2009-09-28 Thread Klaus Darilion
Benny Amorsen schrieb: > Jared Smith writes: > >> Again, the emphasis on the dCAP exam is real-world knowledge of how to >> build a simple small-business PBX with Asterisk. If you've used >> Asterisk in a professional capacity, it should be very straightforward >> to pass the practical portion

[asterisk-users] TE121P Blue Alarm/Recovering

2009-09-28 Thread Klaverstyn, David C
Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzon