[asterisk-users] detecting voltage on fxo

hi is there any way to find out that an fxo module is connected to telco line or not? paradise dove ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://li

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

On Tue, Nov 06, 2007 at 05:16:41PM -0500, [EMAIL PROTECTED] wrote: > I believe that's OpenPBX OpenPBX is a PBX software written in perl by VoiceTronix . I believe you refer to Callweaver. > that tries to derive its timing without > Zaptel devices, however then you need to recompile your Kernel

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

On Wed, Nov 07, 2007 at 07:59:55AM +0100, Hans Feringa wrote: > That was my (mis)understanding as well. It seems that it is currently not > possible to compile the zaptel modules for a 2.6.22 linux kernel. For now > I will not use the trunking option. Zaptel sure can, if you use zaptel 1.4.6 / zap

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

On Tue, Nov 06, 2007 at 03:29:21PM -0500, Anciso, Roy wrote: > Hello list, > > Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I > know there was a bug fix for this but I can't figure out how to select > it. make ECHO_CAN_NAME=OSLEC (after you've applied the patch, that

Re: [asterisk-users] Sangoma S200 and Digium TDM400P together

On Tue, Nov 06, 2007 at 07:53:34PM -0500, [EMAIL PROTECTED] wrote: > What's the result if you do cat /dev/zap ? You mean: cat /proc/zaptel/* But that still won't be good enough. Use genzaptelconf / zapconf included with latest versions of zaptel. -- Tzafrir Cohen icq#168

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

That was my (mis)understanding as well. It seems that it is currently not possible to compile the zaptel modules for a 2.6.22 linux kernel. For now I will not use the trunking option. Thanks, Hans Feringa > On Tue, 2007-11-06 at 18:30 +0100, Hans Feringa wrote: >> I understood that a timing devi

Re: [asterisk-users] dtmf / misdn

This may be what you need: http://www.misdn.org/index.php/FAQ#Why_are_my_dtmf_tones_not_detected_everytime.3F Also, something here may be helpful: http://www.voip-info.org/wiki/view/Asterisk+DTMF#Troubleshooting -jr On Nov 6, 2007 2:12 PM, Hans Witvliet <[EMAIL PROTECTED]> wrote: > Hi all, >

[asterisk-users] wifi

I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with it? I presently use some older Linksys WAP54G APs. I'd like to replace these but in doing so I'd like to be moving in a VOIP friendly direction. I've yet to find a han

Re: [asterisk-users] use dial plan passed arg value in C agi code

On Tue, 6 Nov 2007, Arpit Mehta wrote: > Hi, > > thanks for the replies although i am still confused. The first one is always the hardest :) > On Nov 2, 2007 5:24 PM, Steve Edwards <[EMAIL PROTECTED]> wrote: > > > > The AGI variables are passed via stdin (similar to an HTTP GET request) > > an

Re: [asterisk-users] Mystery phone!

We also sell these phones and ship world wide www.voipperiod.com (See IP0027) Administrator TOOTAI wrote: Kyle Sexton a écrit : Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Not so mysterious: we import those phones in Europe ;-) P

Re: [asterisk-users] MeetMe CPU resources

Just remember if you don't have any Zaptel cards you are going to have to use ztdummy to run app_meetme. Ztdummy essentially requires Linux 2.6, which you should be using anyways. On 11/6/07, Carles Pina i Estany <[EMAIL PROTECTED]> wrote: > > Hello, > > First of all: also thanks to Doug Lytle and

Re: [asterisk-users] Sangoma S200 and Digium TDM400P together

Sorry I mean ls /dev/zap On 11/6/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > What's the result if you do cat /dev/zap ? > > On 11/6/07, Paulo Garcia <[EMAIL PROTECTED]> wrote: > > Hi, > > > > I have these two cards, the Sangoma has 4 fxo interfaces and the > > digium has 1 fxo and 1 fxs. >

Re: [asterisk-users] Sangoma S200 and Digium TDM400P together

What's the result if you do cat /dev/zap ? On 11/6/07, Paulo Garcia <[EMAIL PROTECTED]> wrote: > Hi, > > I have these two cards, the Sangoma has 4 fxo interfaces and the > digium has 1 fxo and 1 fxs. > > After install the sangoma card, my zaptel.conf was configured for that > card. I'm trying to c

Re: [asterisk-users] Everyone is busy/congested: IP Trunk

yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti <[EMAIL PROTECTED]> wrote: > > after a copious loss of follicles :-), I finally got outbound working. > > Basically the channel statement in the call file needs to have the > number to be called. For eg., in test.cal

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

Thanks I was trying to patch 1.4.6 using the 1.4.1.patch. The 1.4.4 patch did the trick:) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Tuesday, November 06, 2007 4:53 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users]

[asterisk-users] Extracting custom headers from SIP REFER

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of th

Re: [asterisk-users] Help: Asterisk info

And why are you asking in the Asterisk list? The absence of that file means you don't have any scsi adapters in your system. Lyle Jarga Jallow wrote: I am getting this error under system info: File Line

Re: [asterisk-users] Mystery phone!

We are in need of an IAX based hard phone. We have used softphones and USB headsets already and they are greatly affected by the other software running on the Windooz laptops and PCs of our users. Does anyone know where we can go to find IAX based hard phones in the US? The one on this link

Re: [asterisk-users] MeetMe CPU resources

Hello, First of all: also thanks to Doug Lytle and Steve Edwards. Just answering one time to all of you. I had the feeling that this computer, for 15 Meetme users, was more than enough... but we wanted to avoid any last-minute surprises! Now we are more sure that everything will work fine. Ah y

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

On 11/6/07, Hans Feringa <[EMAIL PROTECTED]> wrote: > I understood that a timing device (ztdummy if no zaptel hardware is > present) was not necessary anymore with linux kernel 2.6. > > When I enable iax2 trunking I get this warning > chan_iax2.c:8908 build_user: Unable to support trunking on user

[asterisk-users] dtmf / misdn

Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOM

Re: [asterisk-users] Mystery phone!

Kyle Sexton a écrit : > Does anyone know who really makes this phone: > > http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ > Not so mysterious: we import those phones in Europe ;-) POE, 5 accounts, SIP and IAX able, nice audio Good product. -- Daniel ___

[asterisk-users] Pickup Command not working

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EX

Re: [asterisk-users] use dial plan passed arg value in C agi code

Hi, thanks for the replies although i am still confused. my agi script is exten => _.,1,AGI(simple_c_prgm|999); exten => _.,2,NoOp(${MYAGIVAR}); Now i want to set the value of MYAGIVAR to 999 in my c program called simple_c_prgm. This is what I am doing: #include int main(int argc, char *argv

Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID

On 11/6/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: > > > It survives if it goes to a Telus customer, but not if it crosses over > > to Bell, Rogers, etc. > > Well -- here's where you can help me, because our name info is not even > surviving on Telus' own network. I don't really care too much abo

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

Dave Fullerton wrote: > Anciso, Roy wrote: >> Hello list, >> >> Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I >> know there was a bug fix for this but I can't figure out how to select >> it. >> Roy Anciso >> > > You shouldn't need to. As long as you have applied the

Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID

Hi, James -- thanks for your comments. James FitzGibbon wrote: > On 11/6/07, *Stephen Bosch* <[EMAIL PROTECTED] > > wrote: > > We are trying to send caller ID NAME information over a Telus PRI in > Alberta. > > The PRI tech says that he sees the NAME inform

Re: [asterisk-users] Please explain the correct LED color for B410P

[EMAIL PROTECTED] wrote: > Hi. > > > > I have installed B410P in Europe and the cards works more or less ok. My > question is what color should the LED's on the back of the card be when > connected to the PSTN NT box? Is there anywhere some information on the > expected LED color in any given s

Re: [asterisk-users] Asterisk Help

At 13:25 11/6/2007, Jarga Jallow wrote: >Content-class: urn:content-classes:message >Content-Type: multipart/related; > type="multipart/alternative"; > boundary="_=_NextPart_001_01C820AA.C8700D98" > > > >Under asterisk info: Sip registry > >12/12 76.xxx.xxx.

Re: [asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID

On 11/6/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: > > We are trying to send caller ID NAME information over a Telus PRI in > Alberta. > > The PRI tech says that he sees the NAME information, and for calls over > the same network, that NAME info should be reaching the receiving > station, but it

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

Anciso, Roy wrote: > Hello list, > > Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I > know there was a bug fix for this but I can't figure out how to select > it. > > Thanks > > > > Roy Anciso > You shouldn't need to. As long as you have applied the oslec-zaptel

[asterisk-users] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID

We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. The technician was stumped. I suspect there's somethin

[asterisk-users] Selecting OSLEC for zaptel-1.4.6

Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-72

Re: [asterisk-users] 1.4 SIP Jitter Buffer

Are you running the SIP Jitter Buffer? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Luc Moreira > Sent: Monday, November 05, 2007 10:44 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users

Re: [asterisk-users] MeetMe CPU resources

On Tue, 6 Nov 2007, Steve Edwards wrote: > On a dual Intel(R) Xeon(TM) CPU 3.40GHz, all calls are SIP, te410p for > timing only, CentOS 4.5, Kernel 2.6.9-55.ELsmp, Asterisk 1.2.18, MySQL, > various AGI's. According to "top" Asterisk is taking about 80mb and > between 5% and 10% of a cpu. This hos

[asterisk-users] Asterisk Help

Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What c

[asterisk-users] Sangoma S200 and Digium TDM400P together

Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Som

[asterisk-users] Asterisk 1.4 + Presence

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, d

Re: [asterisk-users] MeetMe CPU resources

On Tue, 6 Nov 2007, Carles Pina i Estany wrote: > We would like to have a conference with 15 users aprox. We think that > Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. Your configuration should be more than sufficient. I think (gut feel, no hard stats) I get better audi

[asterisk-users] Help: Asterisk info

I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga <>___ --Bandwi

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

On Tue, 2007-11-06 at 18:30 +0100, Hans Feringa wrote: > I understood that a timing device (ztdummy if no zaptel hardware is > present) was not necessary anymore with linux kernel 2.6. Not quite... this is commonly misunderstood, so let me clarify. Under the 2.6 kernel, ztdummy gets it timing dir

Re: [asterisk-users] Testcall

Ok, muchas gracias, yo soy de Argentina. Estamos en contacto!! - Original Message - From: "Moises Silva" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, November 06, 2007 1:31 PM Subject: Re: [asterisk-users] Testcall Asi es, hablo españo

Re: [asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

Hans Feringa wrote: > I understood that a timing device (ztdummy if no zaptel hardware is > present) was not necessary anymore with linux kernel 2.6. > > When I enable iax2 trunking I get this warning > chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' > without zaptel timi

[asterisk-users] asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking

I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xx' without zaptel timing The linux kernel is 2.6.22-1

Re: [asterisk-users] Everyone is busy/congested: IP Trunk

after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@ And there is no need for a DIAL statement in

Re: [asterisk-users] Queue Statistics Reporting

On 6th November Bob Pierce wrote: Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob Hi Bob You can get free real-time queue statistics from www.orderlyq.com. Just click 'sign-up' button to connect. Rgds Russell _

Re: [asterisk-users] MeetMe CPU resources

In article <[EMAIL PROTECTED]>, Carles Pina i Estany <[EMAIL PROTECTED]> wrote: > > Hello, > > We would like to have a conference with 15 users aprox. We think that > Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. > > We wonder if somebody has some other experience, good

Re: [asterisk-users] Testcall

Asi es, hablo español, soy de México. Anyone interested in R2 support for Asterisk can find more information at: http://www.moythreads.com/astunicall/ - Moy On 11/6/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Ok, Moy, Thank you for your time!!! > Speaking spanish?? > > Cristian. > >

Re: [asterisk-users] MeetMe CPU resources

Carles Pina i Estany wrote: > Hello, > > We would like to have a conference with 15 users aprox. We think that > Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. > > I'd consider that over-kill (Means it'll work fine). We've used a Pentium 3 866mhz with 512mb memory and h

[asterisk-users] MeetMe CPU resources

Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -

Re: [asterisk-users] Queue Statistics reporting

We offer a very comprehensive reporting and real-time monitoring commercial solution called QueueMetrics that has also a free mode for smaller CCs and hobbysts and scales well to multi-server setups with hundreds of live agents. See http://queuemetrics.com As a completely free alternative,

Re: [asterisk-users] Testcall

Ok, Moy, Thank you for your time!!! Speaking spanish?? Cristian. - Original Message - From: "Moises Silva" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 05, 2007 5:42 PM Subject: Re: [asterisk-users] Testcall > You have other

Re: [asterisk-users] Linksys SPA-941 Unavailable

Post the relevant configuration files we'd be glad to help. On 11/6/07, Kim Joung-il <[EMAIL PROTECTED]> wrote: > Hello! > > We are using several Linksys SPA-941 in our office. After IP change occur > devices seems not to be reachable, actually unavailable! Devices is > connected, e.g. we can plac

Re: [asterisk-users] How to delete voice mail messages?

On Mon, Nov 05, 2007 at 12:47:52PM +0100, Michiel van Baak wrote: > On 12:15, Mon 05 Nov 07, voip crazy wrote: > > Hello all, > > > > Could I create a script to delete the first messages on my voice mail? In > > this script should I update any "messages index file" or there isn't any > > file to

Re: [asterisk-users] 7960 Queue Issue

On Mon, Nov 05, 2007 at 12:09:48PM +1100, Nick Brown wrote: > Thanks Eric, this is the case. A bit of a shame that it removes the > functionality for the member to see calls that have not come from a queue > however there is not much choice in the matter. It works for me... somehow... I have Cisco

Re: [asterisk-users] Recording just first part of call?

If you're up to using the Manager interface and your programming language of choice, you could poll the list of active calls and stop recording when their duration exceeds a minute. According to my docs, res/res_monitor.c implements manager commands that could be used to halt current recordings. T

Re: [asterisk-users] Linksys SPA-941 Unavailable

On 11/6/07, Kim Joung-il <[EMAIL PROTECTED]> wrote: > > Hello! > > We are using several Linksys SPA-941 in our office. After IP change occur > devices seems not to be reachable, actually unavailable! Devices is > connected, e.g. we can place a call using SPA-941 but can not receive any > calls...

Re: [asterisk-users] Asterisk and Grandstream both behind different NAT

On 11/6/07, Vivek Shrivastava <[EMAIL PROTECTED]> wrote: > > Hi, > > i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I > have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have > forwarded ports on both Grandstream and Asterisk sides, and using those > ports

Re: [asterisk-users] Recording just first part of call?

Am Dienstag, den 06.11.2007, 11:49 + schrieb Tony Mountifield: > I know that I can record the contents of a call by calling Monitor() > or MixMonitor() from the dialplan just before invoking Dial(). > > I have a potential customer who wants only the first minute of each > call recorded (for id

Re: [asterisk-users] Kirk IP600/3 Wireless Server SIP config

> "RB" == Remco Barendse <[EMAIL PROTECTED]> writes: RB> Did you manage to make a dump of a working configuration from the RB> IP600/3? RB> Would be really useful, can't seem to get it to work properly. Ok, there is one at http://amorsen.dk/complete-IP1200-0f-07-eb.txt. I'm not sure it's ver

[asterisk-users] Recording just first part of call?

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete c

[asterisk-users] Asterisk and Grandstream both behind different NAT

Hi, i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have forwarded ports on both Grandstream and Asterisk sides, and using those ports on Grandstream for SIP and RTP with random ports =no. This setup is wo

[asterisk-users] Linksys SPA-941 Unavailable

Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... Kim __ Do You

Re: [asterisk-users] Asterisk & OpenVZ

On Mon, Nov 05, 2007 at 08:10:33PM -0600, JR Richardson wrote: > Hi All, > > I've got debian (etch), openvz and asterisk up and running using the > openvz wiki guides. The examples use `apt-get install asterisk` and > this will install 1.2.13. Has anyone gotten an VPS to compile the > latest ver

Re: [asterisk-users] Asterisk & OpenVZ

> I've got debian (etch), openvz and asterisk up and running using the > openvz wiki guides. The examples use `apt-get install asterisk` and > this will install 1.2.13. Has anyone gotten an VPS to compile the > latest versions from source? > No problem -- we're running the latest 1.4.x in mu

Re: [asterisk-users] PRI dialout problem with some numbers...

Carlos Chavez wrote: > I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. > This is really the first server I have used with PRI in Mexico as we > normally use MFC/R2. Everything seems to be working except that some > numbers always seem to be busy when you dial them. All th