Re: [Asterisk-Users] Using a TE405P to connect to an existing PBX

Somewhat. You got the remote site right. I have several Voice T1's at my main location, and it runs into a Cisco router which converts it to SIp and sends it to Asterisk. I would like to be able to push certain incoming phone numbers across IAX to another Asterisk server at a remote site. There

[Asterisk-Users] Inbound Free World Dialup - extension not ringing?

  Hi to all the * people out there,   Please kind to me as I am both new to Asterisk and to Linux – But I am learning fast.   My config is quite simple, I’m just following examples and the Wiki:  I have two PC’s running X-Lite phones, these work without problems between each other, and

[Asterisk-Users] Do you speak Czech?

Continuing the work done on Internationalization of Asterisk, we've begun working on Czech. If you speak Czech, we need you help in continuing the work that has been started. Roll up your sleaves and visit http://bugs.digium.com/bug_view_page.php?bug_id=0002013 Thank you! /O ___

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004, Peter Svensson wrote: > On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > > > I'll most likely use a BRI. Do you think this will help to avoid echo? > > Using a BRI will eliminate echos from the pstn connection. Not necessarily! When you call an analog phone via isdn, the

[Asterisk-Users] chan_oh323 loading error

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 l

Re: [Asterisk-Users] Vlan question

There is a way to ensure traffic prioritisation...but it can work out a little expensive. 1. Use 3Com 4400 PWR as your switch. 2. Use 3Com NJ200/NJ220 (US) or NJ205/NJ225 (EU) POE Multiport switches 3. Use 3CNJVOIP-CPOD POE --> 7960 POE/Data splitters for power and data connections to the p

Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?

Thanks!!! That works. A bit cumbersome but at least it works. Why does putting someone on hold and then taking them off again make the ATA (or *) recognize the # key as transfer? Thankyou for your help On Sun, 15 Aug 2004 11:49:50 +0800, MPlus <[EMAIL PROTECTED]> wrote: > For blind transfers, pr

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004 12:15:20 +0200 (CES), Tobias Jönsson <[EMAIL PROTECTED]> wrote: > On Sun, 15 Aug 2004, Peter Svensson wrote: > > On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: > > > > > I'll most likely use a BRI. Do you think this will help to avoid echo? > > > > Using a BRI will elimina

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004, Francis Augusto Medeiros wrote: > Is this small delay annoying enough? Can it be perceived by the part > at the pstn side? Does it disturb fax signals, for example? The echo described by Tobias (originating at the pstn connected user) should only affect the isdn connected pa

[Asterisk-Users] Sip to Sip Calls via Asterisk

Hi All, I have a weird problem. I have asterisk setup using the G729 Codec to receive Incoming calls both from a SIP Gateway (SER and Quintum) and via ISDN using i4l and have rules setup in extensions.conf for sending calls out either back via the SIP Gateway or ISDN. What I want to do is have

Re: [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2

On Sat, 14 Aug 2004 15:04:54 -0700, Vikas Deolaliker <[EMAIL PROTECTED]> wrote: > > I read a few discussions on installing Zaptel modules in Fedora Core 2 with > 2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am > still unable to install by FXO card in my pbx box because the

Re: [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2

I have had success with this using both the X100p (wcfxs and wcfxo) and TE410p (wct4xxp) under Redhat FC2 2.6.5. The instructions are on the wiki, do the following: ln -s /lib/modules/2.6.5-1.358/build linux-2.6 cd zaptel make clean make linux26 make install Having said that I have found the

Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

You need a defination for the inbound FWD and what to do with that.   In my extensions.conf, I have:   [globals] FWDNUMBER=123456 ; your actual fwd number FWDCIDNAME='My Name' FWDPASSWORD=myfwdpasswd FWDRINGS=sip/office FWDVMMBOX=1010   [fwd_out] exten => _123.,1,SetCallerId,${FWDCIDNAME}  ;

Re: [Asterisk-Users] chan_oh323 loading error

You might try setting "P_PTHREADS=1" in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: > I have compiled chan_oh323 and when starting * I get the following. > > [chan_oh323.so]Aug 15 12:40:00 WARN

Re: [Asterisk-Users] chan_oh323 loading error

Hi all! Any one that could give me some input on the problem below? regards Krystian Krystian Filiks wrote: I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.s

Re: [Asterisk-Users] chan_oh323 loading error

Would the command "make P_PTHREADS=1 opt" do the job? Krystian Ryan Wilkins wrote: You might try setting "P_PTHREADS=1" in your Makefile. I'm not actually certain if this will work, but it can't hurt anything. Ryan Wilkins On Sun, 15 Aug 2004, Krystian Filiks wrote: I have compiled chan_oh323 an

Re: [Asterisk-Users] CCM <->(H323) <-> *

My hack worked for me, and still does and last time I checked was still needed. There's no warranty for anyone else. It's possibile there's a cleaner way to fix it, but I've not found it. It was a one line addition to the OpenH323 library source that chan_h323 links against - you don't modify Aste

Re: [Asterisk-Users] Howto remove digits from a called number

Greg Hill a écrit : On Sat, 14 Aug 2004, administrator tootai wrote: Hi list, I have SIP clients and H323 GK connected through h323 channel (Nufone). In h323 conf I gave prefix=09 so all call starting with this prefix are send to asterisk. The context is also given their as [fromh323] But now, i

Re: [Asterisk-Users] chan_oh323 loading error

No. Look in the "Makefile" of the oh323 driver source. Search down through the file for "#P_PTHREADS=1" and remove the #. Then recompile. See if that helps your situation any. It may.. or it may not. On Sun, 15 Aug 2004, Krystian Filiks wrote: > Would the command "make P_PTHREADS=1 opt" do the

Re: [Asterisk-Users] Free MOH MP3

CVS has them - Original Message - From: Wiley E. Siler <[EMAIL PROTECTED]> Date: Sat, 14 Aug 2004 16:50:43 -0700 Subject: [Asterisk-Users] Free MOH MP3 To: [EMAIL PROTECTED] Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answe

Re: [Asterisk-Users] Help - is voip good for in-house calls?

Hi Francis, Francis Augusto Medeiros wrote: Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream B

Re: [Asterisk-Users] Vlan question

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 > - Original Message - > From: [EMAIL PROTECTED] > To: [EMAIL PROTECTED] > Sent: 13 Aug 04, 2:38 PM > Subject: [Asterisk-Users] Vlan question > > Hi, > > I am setting up an Asterisk system with Cisco 7960 phones. I have a PoE > injector to ins

Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?

>From ethereal traces, I noticed that the HT-286 will only send out DTMF signals if it's the calling party, so the act of flashing somehow makes it think that it is now the calling party, so DTMF signals were sent. Not sure if that is a bug or a feature with a certain design consideration. I discov

Re: [Asterisk-Users] Free MOH MP3

Wiley E. Siler <[EMAIL PROTECTED]> wrote: [...] > Does anyone know where I can get some royalty free, cost free music > for my music on hold? The stuff at www.zongoftheweek.com is CC-licensed so should be fair game. Whether you want to inflict some of it on callers is another matter :) -- PGP ke

Re: [Asterisk-Users] chan_oh323 loading error

I have searched through the Makefile and the configure files and could not find any instance of P_PTHREADS. Should I put it there? in that case where? Ryan Wilkins wrote: No. Look in the "Makefile" of the oh323 driver source. Search down through the file for "#P_PTHREADS=1" and remove the #. Then

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004 13:03:49 -0300, Nicolas Gudino <[EMAIL PROTECTED]> wrote: > Hi Francis, > > If you already have the analog telephone wiring in place, and you are on > a budget, I recomend you to use sipura spa-2000 adapters. They are a > whole lot better than GS phones. You can have 3way confe

Re: [Asterisk-Users] chan_oh323 loading error

I think that I found it, I'm compiling PWLIB with "./configure --with-pthreads" Do you think this would do it? Krystian Krystian Filiks wrote: I have searched through the Makefile and the configure files and could not find any instance of P_PTHREADS. Should I put it there? in that case where?

Re: [Asterisk-Users] chan_oh323 loading error

Maybe you are not running the latest version of oh323.. I'm running 'asterisk-oh323-0.6.3a'. On Sun, 15 Aug 2004, Krystian Filiks wrote: > I have searched through the Makefile and the configure files and could > not find any instance of P_PTHREADS. > > Should I put it there? in that case wher

Re: [Asterisk-Users] chan_oh323 loading error

Out of my league.. It may.. You can always try it. On Sun, 15 Aug 2004, Krystian Filiks wrote: > I think that I found it, I'm compiling PWLIB with "./configure > --with-pthreads" > > Do you think this would do it? ___ Asterisk-Users mailing list [EM

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sunday 15 August 2004 12:03, Nicolas Gudino wrote: > If you already have the analog telephone wiring in place, and you are on > a budget, I recomend you to use sipura spa-2000 adapters. They are a > whole lot better than GS phones. You can have 3way conferences and > attendant transfers. With GS

[Asterisk-Users] New $89 VOIP phone

Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice -- Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listin

Re: [Asterisk-Users] SIP connections do not hang up

Hello all, I also have this SIP CANCEL problem and have inverstigated the problem a bit but am not sure if the problem lies in the sipgate proxy or asterisk: 1.) This only happens when you CANCEL an INVITE (obviously) the INVITE is shown below. 2.) sipgate sends a 183 Session Prgress response me

[Asterisk-Users] GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)

Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to mo

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004 13:39:10 -0400, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > Why on earth would you install SPA-2000s and endure that wiring mess? An FXS > channel bank and a BIX strip will save you YEARS in lost time due to wiring > and general messiness! Hello Andrew! I'm sorry to ask th

Re: [Asterisk-Users] chan_oh323 loading error

Ryan Wilkins wrote: Maybe you are not running the latest version of oh323.. I'm running 'asterisk-oh323-0.6.3a'. That is what I'm trying to get going as well with openH323 1.13.5 and pwlib 1.6.6 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Asterisk MIBS

Title: Asterisk MIBS Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing www.faino.org, but this site doesn’t seem to exist anymore, and The Book v2 doesn’t have any references to MIBS. Any pointers greatly appreciated.

RE: [Asterisk-Users] Free MOH MP3

William Suffill [EMAIL PROTECTED] top-posted: > CVS has them > That hasn't been established yet, to my knowledge. The music in CVS appears to have come from a source that doesn't allow free commercial use (www.freeplaymusic.com, according to the CREDITS file). Music on hold is classed as commerc

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sunday 15 August 2004 13:50, Francis Augusto Medeiros wrote: > I'm sorry to ask this really, reeally newbie thing, but... > what would be an FXS channel bank, and where would I find more info > about some popular models? And the same question goes to... BIX > strips! What are those?? :)

Re: [Asterisk-Users] Free MOH MP3

That could right don't really use MOH much but I noticed there was in CVS. Although why would it be in CVS of asterisk if not used for MOH though? On Sun, 15 Aug 2004 18:57:39 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote: > William Suffill [EMAIL PROTECTED] top-posted: > > CVS has them > > > That

[Asterisk-Users] asterisk with InPhonex?

hello has any one got asterisk to work with InPhonex?  if so can you send me your conf information? we are having some problems getting ours up and running. my friend is helping me get it set up.  thanks hank My Inbox is protected by SPAMfighter415 spam mails have been blocked so far.Download

RE: [Asterisk-Users] Free MOH MP3

William Suffill [EMAIL PROTECTED] lazily top-posted: > That could right don't really use MOH much but I noticed there was in > CVS. Although why would it be in CVS of asterisk if not used for MOH > though? > That's the part that needs clarification. Perhaps the music has been included to help pe

[Asterisk-Users] Re: Free MOH MP3

In article <[EMAIL PROTECTED]>, Kevin Walsh <[EMAIL PROTECTED]> wrote: > > I would advise against using the supplied music as MoH until the > permission to do so has been shown. I've just submitted bug #2255 concerning this very point, as that seems to be the best way to get action on something.

[Asterisk-Users] Re:Re:7960 help

- Original Message - > > Message: 2 > Date: Sat, 14 Aug 2004 20:48:55 -0700 (PDT) > From: Gonzalo Gasca Meza <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] 7960 help > To: [EMAIL PROTECTED] > Reply-To: [EMAIL PROTECTED] > > --0-1412799478-1092541735=:62359 > Content-Type: text/plain;

[Asterisk-Users] __use_ast_pthread_create_instead__

Please anyone, When I start * after installing the asterisk-oh323-0.6.3a I get [chan_oh323.so]Aug 15 22:36:44 WARNING[1076252800]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 22:36:44 WARNING[1076252800]: loa

[Asterisk-Users] Internal Distinctive Ringing + Caller ID

I have set up my asterisk PBX to provide a double-ring for outside calls, and a single ring for station-to-station. (I'm talking about ZAP stations in this email). I had to go into one of the .c files and tell it to expect the Caller ID between the 2nd and 3rd rings in order to get the double-rin

Re: [Asterisk-Users] GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)

administrator tootai a écrit : Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything

Re: [Asterisk-Users] Asterisk MIBS

Alagalah wrote: Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing ___www.faino.org_ , but this site doesn’t seem to exist anymore, and The Book v2 doesn’t have any references to MIBS. Any pointers greatl

Re: [Asterisk-Users] Asterisk MIBS

Alagalah wrote: > Hi, > > I was wondering if there are any Asterisk MIBS (specifically regarding > call information) ? > > I noticed a post citing www.faino.org, but this site doesn't seem to > exist anymore, and The Book v2 doesn't have any references to MIBS. > > Any pointers greatly appreciat

[Asterisk-Users] no tones detected

maybe this has been covered before but, i can't find it, has anyone had a problem where outside lines can't use number presses like choose extensions but inside lines can, I am using voicetronix hardware with asterisk and when i call from a station port I hear my greeting and can dial an extension

Re: [Asterisk-Users] no tones detected

On Sun, 15 Aug 2004, Johnathan Bunn wrote: > maybe this has been covered before but, i can't find it, has anyone > had a problem where outside lines can't use number presses like choose > extensions but inside lines can, > I am using voicetronix hardware with asterisk and when i call from a > stat

RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

Hi Lyle,   Thank you so much for your help, I think your information points to using IAX2 rather than registering with FWD from the sip.conf   I have made an attempt to understand this, added the appropriate information into iax.conf, remove old info from sip.conf, gone to fwd and tick

Re: [Asterisk-Users] no tones detected

that just clicked thank you the sample app they included had some freqs set to listen for keypresses I wonder if something is off over there, must be voicetronix's fault On Sun, 15 Aug 2004 15:44:03 -0600 (MDT), Greg Hill <[EMAIL PROTECTED]> wrote: > On Sun, 15 Aug 2004, Johnathan Bunn

[Asterisk-Users] Teliax TOS copied from Vonage?

TelIAX, one of the new VoIP-to-PSTN gateway providers, has their terms of service posted on their signup page: http://teliax.com/user_admin/signup/s1.php They look strangely familiar--it's exactly the same as http://www.vonage.com/features_terms_service.php with s/Vonage/Teliax/. (And it's cut

Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

I have my server on a public IP address, so I am not behind any NAT.   But take a look at http://www.voip-info.org/wiki-Asterisk+config+iax.conf   One of the problems could be the notransfer option in this config file.  I suspect that what you may be seeing(and I am no expert, but have been r

RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing?

IAX2 uses udp port 4569, so you’ll probably have to open that up on your firewall/router. http://www.voip-info.org/ is a good starting place for any asterisk problems - specifically: http://www.voip-info.org/wiki-Asterisk+firewall+rules http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD

Re: Re: [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?

Thanks for the workaround! Hmm...I think it is an unwanted feature. I hope the GrandStream developers "fix" this soon. The workaround isn't something that end-users are going to remember easily or like. On Mon, 16 Aug 2004 00:08:40 +0800, MPlus <[EMAIL PROTECTED]> wrote: > From ethereal traces,

[Asterisk-Users] Newbie with missing .conf files

Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/   I get notices when I try to run asterisk like: parking.conf is depreciated in favour of features.conf Please rename it.  (I'm not getting rename parking .conf to features.conf ???)   I am also missing: m

[Asterisk-Users] 123 Basic configuration files

I need to find some basic configuration files.  Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports?     Don Moskaluk [EMAIL PROTECTED] www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home  

RE: [Asterisk-Users] 123 Basic configuration files

Best starter examples http://www.automated.it/guidetoasterisk.htm   Documentation http://www.digium.com/index.php?menu=documentation   Asterisk will make sample files for you... read teh doucmentation at the first link I listed...   Regards, Wiley            I need to find some basic configur

Re: [Asterisk-Users] Newbie with missing .conf files

Don Moskaluk wrote: Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/ There is a file in /etc/asterisk named 'parking.conf'. It needs to be renamed to 'features.conf'. It's really easy, just "mv /etc/asterisk/parking.conf /etc/asterisk/features.conf" I get no

Re: [Asterisk-Users] Help - is voip good for in-house calls?

On Sun, 15 Aug 2004 13:58:58 -0400, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Sunday 15 August 2004 13:50, Francis Augusto Medeiros wrote: > > I'm sorry to ask this really, reeally newbie thing, but... > > what would be an FXS channel bank, and where would I find more info > > about

Re: [Asterisk-Users] Asterisk MIBS

Alagalah wrote: Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing www.faino.org , but this site doesn ’ t seem to exist anymore , and The Book v2 doesn ’ t have any references to MIBS. I compiled this

Re: [Asterisk-Users] Help - is voip good for in-house calls?

Andrew Kohlsmith wrote: On Sunday 15 August 2004 12:03, Nicolas Gudino wrote: If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant tr

[Asterisk-Users] how can i config a Cisco IAD 2430 config as a sip client

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use

RE: [Asterisk-Users] 123 Basic configuration files

John Todd also has a couple of good articles that should push in the right direction. Check out http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html Best of luck! DC --- "Wiley E. Siler" <[EMAIL PROTECTED]> wrote: > Best starter examples > http://www.automated.it/guidetoasterisk.htm >

[Asterisk-Users] Modified Prepaid doesn't update the balance

Hi all, Has anyone successfully patch the modified prepaid application to update the balance on the card after a call? Best regards, Glynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNS

[Asterisk-Users] consultative transfer with zaptel

Ist there any possibility to use the funktion "consultative transfer"? ( have 2 ISDN-pones attached to the hfc-nt card, configured as zap) With the "#"-key it ist possible to park the call or to make a "blind transfer" at the moment. I have activated threewaycalling in the zapata.conf file: ; int