Re: [Asterisk-Users] 2 Linphones communicating through Asterisk? Solved! Thanks Jon!

On Thu, 2004-03-04 at 15:06, Jon Shamash wrote: quote It looks like you've made a typo in your extensions.conf Doh! What a silly mistake. Yeah, it works now. Thank you very much! Regards Thomas Hi... Being very new to A* myself I understand your fustrations with the manuals

Re: [Asterisk-Users] Best Budgetone firmware?

fine) 3. How to place (naming,config) the files in tftpdir Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] BCM Wireless SIP Phone

Hi, Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm Regards, Steven Thomas Network Integration Services IBM Australia Ph: 0404 099 262 NH011, IBM Centre, 601 Pacific Hwy, St Leonards, 2065.

AW: [Asterisk-Users] who has German voice files ?

Wait a week and you can have german files from one of our customers, who wants to donate such files. Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jakob Strebel Gesendet: Donnerstag, 11. Marz 2004 14:31 An: [EMAIL PROTECTED

[Asterisk-Users] RTP Read error: Resource temporarily unavailable (DTMF Issues)

2 codecs are set to PCMU and PCMA (tried to switch those arround too). Any help very appreciated! Thanks, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

AW: [Asterisk-Users] AGI script will not be terminated

An: Asterisk-Users Mailing-list Betreff: Re: [Asterisk-Users] AGI script will not be terminated Hi Thomas, Do you answer correctly in your dial plan ? I met the same problem but unfortunately I don t remember what I was doing wrong :/ exten = 1112,1,Answer exten = 1112,2,Wait(1) exten = 1112,3,agi

Re: [Asterisk-Users] Re: RTP Read error: Resource temporarily unavailable (DTMF Issues)

Tony Mountifield wrote: In article [EMAIL PROTECTED], Thomas Gallaway [EMAIL PROTECTED] wrote: Hi I am working on this since a while now and seem to be stuck. Here is my issue: I have a bunch of Budge Tone 101's. Asterisk is set up. 4 Incoming PSTN lines. It all works fine just the DTMF

[Asterisk-Users] Pulver WiSIP Dual Line and Hold?

Hi, I have received my WiSIP phone - works well for basic functions of call answer and hang-up! Does anyone know how to enable Dual line support, Hold and Transfer functions with this phone via Asterisk. Thanks, Regards, Steven Thomas

Re: [Asterisk-Users] High latency from Europe, 500-800ms.

128ms pings to europe (germany). -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] High latency from Europe, 500-800ms.

Angel Gabriel wrote: I'm no expert on * , I don;t even think i class as a newbie yet, but my understanding of a sat. link to the net is that the link is one way, as in downstream, and you still need an upstream conenction to your service provider. If this is the case, then remember, your

[Asterisk-Users] Registration from xxx failed for 'xxx'

) 255.255.255.255 0Unmonitored 113/113 192.168.1.113 (D) 255.255.255.255 5060 Unmonitored Thanks for any input on this. -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Registration from xxx failed for 'xxx'

for that port. On Fri, 2004-03-19 at 16:26, Thomas Gallaway wrote: Hi Here is my problem. I have 2 phones (Grandstream Budge Tone-100) loosing the sip registration every 4 hours. I can not find out why. It seems like the registration fails, then a few minutes after registers sucessfull. Mar 19 14:06

Re: [Asterisk-Users] asterisk installation problem

OK, I solved the problem by myself: openssl-devel was not installed. Unfortunately, there's not a deb-package, so I had to convert the RPM. Regards, thomas On 21 Mar 2004 at 21:29, Thomas Schroeter wrote: Hello, I have the following problem installing Asterias on Debian woody

[Asterisk-Users] Asterisk Memory Usage

I am watching memory usingTop and when Asterisk is running, memory usage increases 8 k every 10 seconds or so. I stop Asterisk and this memory increase stops. This doesn't sound normal. Anybody else experiencing this?

[Asterisk-Users] Asterisk Possible Memory Leak

Sorry, my previous post was almost unreadable. I am watching memory using Top and when Asterisk is running, memory usage increases 8 k every 10 seconds or so.  I stop Asterisk and this memory increase stops.  This doesn't sound normal.  Anybody else experiencing this?

Re: [Asterisk-Users] Two AVM Fritz Card (hack does not work) what I am doing wrong?

Hallo Jakob Strebel On Mon, 22 Mar 2004 18:27:44 +0100 you wrote: Hi, I tried to install the following hack. http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO But the 2nd AVM Fritz PCI card is still not showing up. My environment is: debian 2.4.24 (asterisk 0.72) Just a quick

Re: [Asterisk-Users] 10 day old email, virus already received

[EMAIL PROTECTED] wrote: Hi, I´m a member from this list and I do not have a virus because I check my email from my server linux. cheers. vozip Mensaje citado por randulo [EMAIL PROTECTED]: For info, I receive the mailing list on a brand new account that is not used for anything else.

[Asterisk-Users] s extension don't work on TDM40B

up the phone the s extension will be never executed. Whats wrong ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL

[Asterisk-Users] Problem with outgoing spool...

when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the i4l driver have a bug ? Or maybe the spooler is defective ? Thanks for Help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18

AW: [Asterisk-Users] Problem with outgoing spool...

* and my script. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 13. Juni 2003 15:23 An: Asterisk User Betreff: [Asterisk-Users] Problem with outgoing spool... Hi all, i 've written a little Callgen script

AW: AW: [Asterisk-Users] Problem with outgoing spool...

PROTECTED] Betreff: Re: AW: [Asterisk-Users] Problem with outgoing spool... Hi Thomas, what about file names? Do all calls (in every run) get new filenames? I had some problems when the file names remained the same. * wouldn't detect the file as new. I added -timestamp to the file names and everything

Re: [Asterisk-Users] Cisco 7960 config?

's line selection screen, add this entry: line1_shortname: Asterisk Test As my first contribution to the list, I hope this helps! Respectfully, Thomas A. Roberts Seegence Corporation www.seegence.com Quoting Dave Weis [EMAIL PROTECTED]: I finally got the power supply for my 7960 and am

[Asterisk-Users] number of digits from incoming msn on i4l modem

. Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] where to get adsi phones in europe ?

Hi all, have anybody an idea where to get adsi phones in europe ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] help with pri configuration..

: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

AW: [Asterisk-Users] help with pri configuration..

, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Montag, 23. Juni 2003 11:34 An: Asterisk User Betreff: [Asterisk-Users] help with pri configuration.. Hi all, can somebody help me with pri configuration? Here my

[Asterisk-Users] NoOp gives an ringing indication ?

[tel1] exten = s,1,GotoIf($[${FREE1} = 1]?s|4:s|2) exten = s,2,Playback,gesperrt exten = s,3,Hangup exten = s,4,NoOp exten = _X.,1,Dial,Modem/g1:${EXTEN} exten = _X.,102,Hangup What can i do ? Thanks for help, Thomas. *** beroNet technologies GmbH

AW: [Asterisk-Users] parsing bug? (using PGSQL)

OK. I see. This works. Thank you, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Martin Pycko Gesendet: Dienstag, 24. Juni 2003 18:03 An: Asterisk User Betreff: Re: [Asterisk-Users] parsing bug? (using PGSQL) If you use brackets

AW: [Asterisk-Users] no sound pri -- h323

Hi Michael, i have tried both. H323 and OH323. But meanwhile i've found out that the problem is at the teles-pbx. Thanks, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Mittwoch, 25. Juni 2003 15:49 An: [EMAIL

[Asterisk-Users] Congestion or Busy app using I4L indicate ringing

? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] how to identify user using chan_H323

party says ERROR[16401]: File chan_h323.c, Line 963 (setup_incoming_call): Call from user 'root' rejected due to no default context even though i gave thaeger as username? Thanks for help, Thomas. ___ Asterisk-Users mailing

[Asterisk-Users] bug in cdr ?

after entering s,4,Background the user can dial digits, i think. If the user was connected the called id is s and not the dialed number. Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow

AW: [Asterisk-Users] how to identify user using chan_H323

Why not ? We want use * as Viop gateway to other carriers... And there we have to identify us i think. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jeremy McNamara Gesendet: Donnerstag, 26. Juni 2003 20:26 An: [EMAIL PROTECTED

AW: [Asterisk-Users] bug in cdr ?

Please, can anybody help me with this ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Donnerstag, 26. Juni 2003 19:00 An: Asterisk User Betreff: [Asterisk-Users] bug in cdr ? Hi all, i have a TDM40B

[Asterisk-Users] PGSQL app and pbx parsing :-(

, is there a trick i can use ? Or ist this just a BUG ? Thanks for Help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] PGSQL app and pbx parsing :-(

Please help. this is generally a problem with arguments that contain , or (, i think. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Sonntag, 29. Juni 2003 15:18 An: Asterisk User Betreff: [Asterisk-Users] PGSQL

AW: [Asterisk-Users] PGSQL app and pbx parsing :-(

Hi Adam, i think the real problem is the ,. This will be allways replaced through |. Regrads, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Adam Goryachev Gesendet: Dienstag, 1. Juli 2003 10:12 An: [EMAIL PROTECTED] Betreff: RE

Re: [Asterisk-Users] three way calling and cisco ata 186

Pavel Zheltouhov wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? Thomas ___ Asterisk-Users mailing

AW: [Asterisk-Users] Wildcard E100P resellers in Europe ?

Here you can get the reseller data -- http://www.digium.com/index.php?menu=resellers Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Nicolas Cartron Gesendet: Freitag, 11. Juli 2003 09:53 An: [EMAIL PROTECTED] Betreff: [Asterisk

[Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)

Hi all, i have a E400P in my P III 1,4 GHz machine. When i start the tor2 driver (modprobe tor2) then i can see (with top) that the System takes 20 - 30 % CPU usage. Is this normal ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL

AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)

Please can anybody help me with this, have anybody experiences with the tor2 driver? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Freitag, 11. Juli 2003 13:23 An: Asterisk User Betreff: [Asterisk-Users] mod tor2 takes 20

Re: [Asterisk-Users] mgcp problems

threewaycalling with my ATAs. Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

AW: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)

Steve, thanks for your explanation. This is the cause for the fact that if i change the pci slot, the problem is blown away, i think. Maybe the IRQ sharing is the cause ... Thanks a lot and best regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] Newbie Help

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory

Re: [Asterisk-Users] Newbie Help

# +44(0)7764486175 US# +1(917)4386847 - Original Message - From: Thomas Elliott [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 11:16 AM Subject: [Asterisk-Users] Newbie Help Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make

[Asterisk-Users] h323 gateway call lost after 74sec always

Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered

Re: [Asterisk-Users] h323 gateway call lost after 74sec always

Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router

Re: [Asterisk-Users] h323 gateway call lost after 74sec always

Steven Thomas wrote: Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly

[Asterisk-Users] go on in context after the destination channel hung up?

for preventing the hangup for the originating channel and go on in the current context ? like : exten = 111,1,Dial,Zap/4 (after Zap/4 has hung up) exten = 111,2,whatever ... Any ideas ? Thanks for help, Thomas. ___ Asterisk-Users

[Asterisk-Users] Chan_oh323 Dial format / voice latency 4 to 5 secs

Hi, Can someone confirm the format of the Dial string for a H.323 gateway using chan_oh323? The format I have working is: exten = 5000,1,Dial(OH323/h323:[EMAIL PROTECTED]) I have 5000 as a speed dial - the extension functions, but the voice latency within the call to the analog phone

[Asterisk-Users] go on in current context after destination channels hung up ?

for preventing the hangup for the originating channel and go on in the current context ? Or, is there a way to implement this feature ? I think it make sense to implement such a feature, but i don't know where i can prove. Any ideas ? Thanks for help, Thomas

AW: [Asterisk-Users] go on in current context after destination channels hung up ?

Where i can say n for next ? I can not see an option n in your description in app_dial.c Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Mark Spencer Gesendet: Montag, 28. Juli 2003 15:52 An: Asterisk User Betreff: Re

AW: [Asterisk-Users] go on in current context after destination channels hung up ?

Aehmm... (Mark,) :-) i think i understand now ... i have to make the n option by myself. I'am not the best in the english language and i don't know the niceties in their. Forgive me, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von

Re: [Asterisk-Users] Call transfer on ATA186

?) works with incoming and outgoing calls (with Flash). I never tried H323 because of the mammut sources to compile. With SIP there was no way to get all things working. Sometimes the sound (over ATA) is choppy. Then i have to reboot the ATA and everything is fine again (any hints?). - Thomas

[Asterisk-Users] Chan_h323.so native?

Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? Thanks. Regards, Steven Thomas

[Asterisk-Users] Chan_h323 one way audio

, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Syntax for 2 ISDN Cards

On Tue, Jun 01, 2004 at 11:34:28PM +0200, Gunnar Schaller wrote: ...cut When I try to make 3 simultaneous connections from SIP to ISDN the first and second one works, but on the third connection this happens: -- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in new stack

Re: [Asterisk-Users] Syntax for 2 ISDN Cards

On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote: ...cut chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do

Re: [Asterisk-Users] chan_capi and DDI (Anlagenanschluss)

Hallo Holger Schurig On Mon, 7 Jun 2004 15:17:17 +0200 you wrote: Did you configure your CAPI driver to run in PTP mode? (not PTMP mode) Hmm, I guess you brought me on the rigth track. # cat /proc/capi/controllers/1 name fritz-pci io 0xA800 irq 5

Re: [Asterisk-Users] AVM B1 and PTP mode

Hallo Holger Schurig On Mon, 7 Jun 2004 17:08:53 +0200 you wrote: Hi ! I've fetched a spare AVM B1 card from the cellar, and installed it. After modprobe b1pci I did capiinit and capiinit moaned about a missing t1.b4. Dont use modprobe, use capiinit start ! Configure the

Re: [Asterisk-Users] german localization for mailbox available?

On Mon, Jun 14, 2004 at 06:44:45PM +0200, Frank Sautter wrote: hi, i just wanted to ask if there is a german localization for the audio files of the mailbox available on the net. Shure! Just ask the wiki about voices it might share the secret with you :) regards frank sautter

Re: [Asterisk-Users] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems

On Tue, Jun 15, 2004 at 07:43:32PM +1000, Shaun Ewing wrote: That's the last thing I wanted to hear :-( Apparently my ISDN card (Eicon Diva 2.02 as I mentioned) supports CAPI, but I've only been able to find Windows drivers for it. Have a look at http://www.melware.de Maybe that helps. --

[Asterisk-Users] Asterisk does not start when cdr_odbc ist configured

Hi, i want to load the cdr into oracle using unixODBC. I'm using RH 9 2.4.20-30.9smp, unixODBC 2.2.6, easysoft odbc driver for oracle 1.3.1. My unixODBC is working well. With isql i can connect to the database, do selects, inserts and so on. I created the table cdr as described on the asterisk

[Asterisk-Users] PRI immediate=no

= no. Using the redial option on an analogue phone also works fine, it seems that asterisk does not wait for more digits but answers as soon as the number matches anything in extensions.conf. But shouldn't that be fixed with immediate=no? Regards, thomas

RE: [Asterisk-Users] PRI immediate=no

zapata.conf? Yes, I have. Hope that helps...? Regards, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] capi.so problem on startup

Hallo Tobi Anton On Wed, 23 Jun 2004 12:26:37 +0200 you wrote: Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with asterisk -c I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got

[Asterisk-Users] missing .gsm in VoiceMailMain(2)

Hi Folks I try the following within context: exten = foo,foo,VoiceMailMain After providung MailBoxNumber I get asked for PassWord. If now the input fails I see on CLI Playing 'vm-incorrect' followed by Playing 'vm-password' and I can hear both messages. Next try is: exten =

Re: [Asterisk-Users] missing .gsm in VoiceMailMain(2)

Hallo Jean-Yves Avenard On Tue, 6 Jul 2004 22:08:39 +1000 you wrote: On 06/07/2004, at 10:00 PM, Thomas Niesel wrote: I try the following within context: exten = foo,foo,VoiceMailMain After providung MailBoxNumber I get asked for PassWord. If now the input fails I see on CLI

Re: [Asterisk-Users] HFC- Colongne TE Mode

Hallo Junaid Saeed Uppal On Wed, 7 Jul 2004 15:41:49 +0500 you wrote: Hello There, I am trying to get Asterisk to work with Billion ISDN Adaptor, But i couldnt get isdn4linux to work. I am pretty new with isdn card but this is the only available option here right now , I've looked at this

[Asterisk-Users] chan_mISDN test release....

. Have a lot of fun! Best Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow (bei Berlin) FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] Asterisk don't listen to my phones

lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas DEILLON ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Kernel panic with two Fritz cards

Hi, possible IRQ conflict? Do you see the binary (in the panic dump) which caused the kernel panic? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] No voice call from H.323-phone to SIP-phone

Hi, Did anyone have any comments on the below problem - or did you (shong ching) manage to solve this? I have the same issue - any assistance would be great. Thanks. Regards, Steven Thomas

Re: [Asterisk-Users] Chan_h323 one way audio

not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua

Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323

I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas

Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323

Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas

[Asterisk-Users] Chan_h323 support for phone numbers via gateway?

Does chan_h323 support phone number calling via a gateway? ie., something like calling 5000 forwarded to: exten = 5000,1,Dial(h323/[EMAIL PROTECTED]) if so - what format should the exten be in? Thanks. Regards, Steven Thomas ___ Asterisk

[Asterisk-Users] Chan_h323 does not seem to send the destimation number to gateway

fine if it is just an IP address that it is calling, ie, a softphone. Thanks for your help Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] ADSI Programs

Hi, one question: What you mean with unlocked ? -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von jerk face Gesendet: Mittwoch, 27. August 2003 18:31 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] ADSI Programs I just received an unlocked ADSI

[Asterisk-Users] additional digit in front of the dialed extenesion by outgoing pri/E1 call

anybody help me ? Thank you very much, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] some pri questions...

the * so that * detecting dtmf during a call ? Thanks for answering questions, regards, Thomas. :-) *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL

[Asterisk-Users] Problem with SIP: Maximum retries exceeded

anything to configure like Max retries Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

AW: [Asterisk-Users] some pri questions...

, but you'd have to make some changes to chan_zap.c PSTN channel driver. regards Martin Thanks for answering questions, regards, Thomas. :-) *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328

AW: [Asterisk-Users] Call script after hangup

Hi Frank, why you so complicated ? Try following: [incoming] exten = s,1,Playback,welcome exten = s,2,Record,msgfile:gsm exten = h,1,System(/home/frank/callscript.pl) as sample ... :-) Regards, Thomas. -Ursprüngliche Nachricht-Von: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] IAX sound probs

. What' s wrong? Is there another port wich i have to nat ? Regards, thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL

[Asterisk-Users] CallerID through the GnuGK - does this work?

Hi - can anyone confirm or deny that CallerID works through (passes through) the GnuGK? ie., X100P - Asterisk - GnuGK - Gateway Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

Re: [Asterisk-Users] delay problem in h323

- not sure why! Regards, Steven Thomas andrea [EMAIL PROTECTED

Re: [Asterisk-Users] delay problem in h323

also. I would suggest trying chan_h323 as an alternative. Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its

Re: [Asterisk-Users] delay problem in h323

I assume it manages the signal part of the RTP stream but not the RTP voice stream at the codec level? Maybe someone else can comment on the translation methodologies within Asterisk? Regards, Steven Thomas

[Asterisk-Users] PPP over ISDN BRI (modem_i4l) ?

Hi all, is this possible ? Make an incoming data call with ppp ? (like ZapRas...) Thanks for help, Thomas *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779

[Asterisk-Users] h323 gatekeeper registration failed

or oh323.conf to the ip address from the gatekeeper. gatekeeper=x.x.x.x But no one of the both driver can register to this gateway. Is there another thing that i have to keep ? I need yours help urgently. We want to go online with our *-gateway as soon as possible. Thanks, Thomas

AW: [Asterisk-Users] h323 gatekeeper registration failed

Hi Michael, this gatekeeper works without a password but with a H323-ID, but this will be send with the dial command, i think. Here is the trace with trace level 10 (?) Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael

[Asterisk-Users] no ring tone analog Zap -- I4L

is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XX|60|r so it is no wonder that i never noticed that the ring tone not working Have anybody an idea ? Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing

AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed

Hi Michael, registration is working now, it dials out the phone is ringing but then comes a hang up I'am i lttle newbe on h323 :-) Can you take a look on the log file ? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael

AW: AW: AW: AW: [Asterisk-Users] h323 gatekeeper registration failed

Ahh... you mean it's a codec problem? This can be... I ask my provider :-). If this was not the prob, i would get in touch with you. Regards, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Michael Manousos Gesendet: Donnerstag, 18

[Asterisk-Users] codec probs wit g723.1

GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED

[Asterisk-Users] ringing tone on analog Zap channel question

to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email

[Asterisk-Users] No sound on PSTN -- */PRI

Hi all, i tried to make a call from public pstn in our */E100P. Config is following: exten = _X.,1,Playback(testgsm) But what i hear is one dtmf tone and then nothing... Any ideas ? Regards, Thomas. *** beroNet technologies GmbH Dipl.- Ing

AW: [Asterisk-Users] No sound on PSTN -- */PRI

2003 18:10:37 +0100, Scott Stingel wrote Have you tried starting asterisk with -c? It should give you some detail as to what is happening with the call. Scott M. Stingel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Haeger Sent

RE: [Asterisk-Users] No sound on PSTN -- */PRI

, the sound on the scond one vanishes. My conf: exten = _X.,1,Answer exten = _X.,2,Playback(outofdark) ;(mp3 file) Can somebody help? Thanks, Thomas. -Ursprngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Samstag, 20. September 2003

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