Re: [Asterisk-Users] Getting netmeeting to work with Asterisk

>I've played with modifying the extensions.conf and h323.conf but don't >have things right. I keep getting a message on the >console: > >ERROR[376849]: File chan_h323.c, Line 974 (setup_incoming_call): Call from >user 'Simon' rejected due to no default context > >However I am unsure what this rea

Re: [Asterisk-Users] Modem => Serial ?

> > Actually Dan, you are mistaken here, many serial fax/data/voice modems > > come with answering machine programs which tell the modem to > > send the voice data through the serial line. Also the vgetty > > program in linux does the same. > > The problem is using the AT command set, not whether

Re: [Asterisk-Users] Fixed Cellular adapters/terminals

Hi TC, I was stating my opinion which is why I didn't provide any examples. If you need examples here you go. I have to administer equipment remotely. Since these devices have no network interface this is not possible. I am forced to use a handset and key sequences whenever I need to reprog

Re: [Asterisk-Users] chan_capi with avm c2 only uses one BRI

I can call out of both, but not at the same time On Wednesday 04 June 2003 12:36, T Aksoy wrote: > Are you sure that your telco hasn't busied one of them out? We had a > similar problem. > > - Original Message - > From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> >

RE: [Asterisk-Users] Clock Sync

On Wednesday, June 04, 2003 1:25 AM, Jeremy McNamara [SMTP:[EMAIL PROTECTED] wrote: > There is a little discussion going on comp.dcom.voice-over-ip and I > could not answer these two questions... The author has sorta > clarified his two questions, but they still mean little to me. > >> 6) A redund

RE: [Asterisk-Users] Getting netmeeting to work with Asterisk

In h323.conf did you add sections like [eanders] type=friend context=default incominglimit=6 outgoinglimit=4 You need the context=default Erik > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Simon J Mudd > Sent: Wednesday, June 04, 2003 6:46 AM > To:

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

> > > Some hardware modems are able to do full duplex, and some don`t. They > do > > > predomenantly however have drivers for linux available that allow > > > voice-modem use. > > > > > > Either way - you would then need an asterisk channel for full-duplex > voice > > > modem. > This is the point..

Re: [Asterisk-Users] Modem => Serial ?

Hi, Check: http://www.smlink.com/download/Linux/slmdm-2.7.14.tar.gz This is the modem I talk about Smartlink SM56 USB version This is extracted from the README file: Features Modem: V.92, V.90, V.34, V.32bis, V.23, V.22, V.21, Bell 103/212. Flow control: V.42, MNP 2-4. Compression: V.4

[Asterisk-Users] Budgettone 100 phone Configuration

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://

[Asterisk-Users] X100P creating a short-circuit on line

We have a Digium X100P FXO single-line PCI card installed on our Asterisk test server and it has been working fine. ie. We can initiate and receive PSTN calls and transfer between PSTN and VoIP without any problems. However, we have had a surprised engineer visit from our telco, British Telecom, y

[Asterisk-Users] Getting netmeeting to work with Asterisk

Hello All, Finally I realised that the Asterisk demo setup didn't include support for h323. (Maybe it should have been obvious) so I went to work out how to get the h323 channel running. I had openh323 and pwlib installed as I'd been playing with vocal so it didn't take long to do cd asterisk/ch

Re: [Asterisk-Users] Call Transfer Problem

Sorry, I might be being stupid, but I don't see what the problem is. Following your example, 1. Secretary calls someone for the Boss 2. Other caller answers, Secretary asks other end to wait. 3. Secretary presses the flash button (or recall or whatever it's called on the phone) 4. Secretary dia

Re: [Asterisk-Users] Modem => Serial ?

The problem with the few softmodem drivers I've dealt with is that parts of the driver are _NOT_ in source form. The parts that are in source form are basically a serial interface to the dsp library that does direct hardware access as it would have kernel level privs from having been called from in

Re: [Asterisk-Users] Modem => Serial ?

With a software modem, you cannot bypass AT commands and directly acces the hardware for voice applications, as you have the source code for the modem driver himself? BR, Dan > The problem is using the AT command set, not whether or not serial > communications can carry voice. Remember that T1 i

Re: [Asterisk-Users] chan_capi with avm c2 only uses one BRI

Are you sure that your telco hasn't busied one of them out? We had a similar problem. - Original Message - From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]> Sent: Wednesday, June 04, 2003 10:46 AM Subject: [Asterisk-Us

Re: [Asterisk-Users] Call Transfer Problem

yes, u are quite right, you can find this feature in almost every pbx now.   We are also wondering whether, presently some one is implementing this feature or not, if no body is doing that, we can start on that   Surajee     - Original Message - From: George Lin To: [EMAIL

Re: [Asterisk-Users] Modem => Serial ?

On Wed, 2003-06-04 at 01:49, Anthony Wood wrote: > On Wed, Jun 04, 2003 at 09:05:54AM +0300, Dan wrote: > > Hi, > > > > Serial voice modems use separate jacks for audio in and audio out. > > The audio stream cannot be passed through the serial line. > > Actually Dan, you are mistaken here, many s

[Asterisk-Users] chan_capi with avm c2 only uses one BRI

hi all it seems like whatever I do, I can't use more than 1 BRI on my AVM C2 with chan_capi. Both channels seem to work, but not at the same time. And - yes - they're connected to different NT boxes :) Any ideas? kapejod? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.

Re: [Asterisk-Users] ata186 and 9 for outgoing line type dialplans

You problem is that in SIP, your dialtone is provided by the SIP device. Asterisk is not sending dialtone. Your ATA is doing the dialtone the whole time. What you may want to do to make life really easy is to set up the ata186 to work like a hotline phone. There was comments on the list about this.

Re: [Asterisk-Users] 7940 SIP upgrade

Hi, Check this link: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/sip7960/siprns/phnrn43s.htm 7960 and 7940 are allmost identical (6/2 external lines) BR, Dan - Original Message - From: "argnet" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 04, 2003

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

Hi, > > > As it is a software modem, it can do full duplex. > > > > I think you are confusing the issue here. Question is: can the softmodem > > do full duplex with linux drivers? I don't know yet... I hope to have some time to check. The linux software can do answering machine too (it states t

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

Hi, > > When asterisk is started, I see something like this in the log: > > > > [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem > > Driver) > > I guess it's a specific chipset for certain hardware modems. I have a PCI soft modem based on Conexant V.90 chipset too... > >

Re: [Asterisk-Users] Asterisk Works on Linux on Sparc

[EMAIL PROTECTED] wrote: I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM

[Asterisk-Users] 7940 SIP upgrade

Hi to all, can someone give me step-by-step guide how to upgrade Cisco 7940 phone with SIP firmware (I've got firmware image file *.bin)...so I can use it with my asterisk server Any help appreciated, Victor ___ Asterisk-Users mailing list [EMAIL P

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

> > As it is a software modem, it can do full duplex. > > I think you are confusing the issue here. Question is: can the softmodem > do full duplex with linux drivers? > > Some hardware modems are able to do full duplex, and some don`t. They do > predomenantly however have drivers for linux ava

Re: [Asterisk-Users] Fixed Cellular adapters/terminals

>>Depending upon how you use these devices they are both good and bad. I personally would not recommend >>them to anyone. >>Rob. any experience here, ?? I wish when ppl make comments like this they would give some hard facts and examples ___ Asterisk

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

On Wed, Jun 04, 2003 at 09:15:25AM +0300, Dan wrote: > Hi Woody, > > As it is a software modem, it can do full duplex. > . If it can do full duplex, then no-one has pointed out to me why a driver can't be written for it. > > Also, I don't think Asterisk has drivers for this sort of thing, > > as

Re: [Asterisk-Users] Modem => Serial ?

On Wed, Jun 04, 2003 at 09:05:54AM +0300, Dan wrote: > Hi, > > Serial voice modems use separate jacks for audio in and audio out. > The audio stream cannot be passed through the serial line. Actually Dan, you are mistaken here, many serial fax/data/voice modems come with answering machine program

[Asterisk-Users] Clock Sync

There is a little discussion going on comp.dcom.voice-over-ip and I could not answer these two questions... The author has sorta clarified his two questions, but they still mean little to me. 6) A redundant configuration that can synchronize on, and share one, two, and more network clocking sig

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

Hi Woody, As it is a software modem, it can do full duplex. . > Also, I don't think Asterisk has drivers for this sort of thing, > as most of the hardware is half-duplex. When asterisk is started, I see something like this in the log: [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset)

Re: [Asterisk-Users] Modem => Serial ?

Hi, Serial voice modems use separate jacks for audio in and audio out. The audio stream cannot be passed through the serial line. BR, Dan - Original Message - From: "Gary" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 04, 2003 7:17 AM Subject: [Asterisk-Users] Modem

Re: [Asterisk-Users] Fixed Cellular adapters/terminals

Title: Re: [Asterisk-Users] Fixed Cellular adapters/terminals Depending upon how you use these devices they are both good and bad.  I personally would not recommend them to anyone. Rob. On 6/3/03 19:02, "TC" <[EMAIL PROTECTED]> wrote: FYI. http://www.telular.com/products/index.asp These look

[Asterisk-Users] Modem => Serial ?

Now with the usual discussion which arises with compatible internal modems for FULL DUPLEX voice etc... It really makes me think (again) about serial communications Most external modems will now talk at 115k2 so maybe some will explain when/what/how/etc of why a serial interface could NOT be

[Asterisk-Users] Drivers etc [was] Voice Modem + Soundcard Driver

Actually with all this discussion about modem drivers etc... Maybe, just maybe there is someone out there (Mark ?) who might add a new driver to their wishlist and inform the list. Problems are the actual modem chip MUST support FULL DUPLEX VOICE. Which severely limits you to suitable chipsets &

Re: [Asterisk-Users] Fixed Cellular adapters/terminals

This sounds great. Has anyone actually used one? On Tue, 3 Jun 2003, TC wrote: > FYI. > http://www.telular.com/products/index.asp > > These look like the right solution for any one wanting a cellular FXO device > for * > that interface with the digital cellular networks (GSM ,CDMA etc) > similar

[Asterisk-Users] ata186 and 9 for outgoing line type dialplans

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXX,1,Dial(IAX2/[EMAIL PROTEC

[Asterisk-Users] Fixed Cellular adapters/terminals

FYI.http://www.telular.com/products/index.aspThese look like the right solution for any one wanting a cellular FXO devicefor *that interface with the digital cellular networks (GSM ,CDMA etc)similar to the www.cellsocket.com or the mystery FCT's http://www.ericsson.com/products/products_az

Re: [Asterisk-Users] Voice Modem + Soundcard Driver

On Tue, Jun 03, 2003 at 11:09:28AM +0300, Dan wrote: > Hi, > > I have an USB voice modem which does not need to be connected to the > soundcard in order to build an answering machine. > It appear in the system as another sound card. > I have tested a free Answering machine application and it works

[Asterisk-Users] Asterisk localization

Hi All,   I’ve been working with asterisk for about two months, and I would like to contribute to the project on the localization side, mostly making it easier to translate text output and pre-recorded messages.   My goal is to discuss with you guys a framework for localization/transla

Re: [Asterisk-Users] Asterisk Works on Linux on Sparc

[EMAIL PROTECTED] wrote: chan_h323.so will not load too. It gives undefined symbol error. This is typical version skew. You have to make clean opt open h323 and pwlib and make clean install chan_h323, but I think there are gonna be other issues with the Makefile... Like exactly which lib to l

Re: [Asterisk-Users] Example of the Transfer application?

The transfer application generates the flash on analog interfaces. It won't work w/SIP. Martin On Tue, 3 Jun 2003, John Todd wrote: > > OK, I'm stumped. I have no idea how one would use the Transfer > application. Perhaps it is because I am an all-SIP environment, but > I don't see what purpos

[Asterisk-Users] Initial Connection Hangup (T100P) and Ringing Failure

I ran into a couple of odd problems with a new Asterisk box I have set up... I have a T100P connected to a channelized T-1 (24ch) to the PSTN. The T-1 is set up properly so far as I know and seems to work perfectly with one exception: When a call comes in, asterisk starts a 'simple switch' for it,

[Asterisk-Users] Example of the Transfer application?

OK, I'm stumped. I have no idea how one would use the Transfer application. Perhaps it is because I am an all-SIP environment, but I don't see what purpose it solves (maybe I'm not thinking in the right mindset.) My understanding of Dial is that the only way to escape is via a hangup or pres

[Asterisk-Users] Sound: Recording overrun

Hello All, I've just been made aware of Asterisk and have installed it. Things seem to be working: I can dial from the console the internal numbers and hear the answerphone messages and do other interesting stuff. however when I hangup the call I see a continuous stream of messages saying "Sound

Re: [Asterisk-Users] Detect hangup on unanswered POTS call

On Tuesday 03 June 2003 12:17, Nathan Lutchansky wrote: > I've been using * at home for a while now and I'm quite happy with > how it works. Having voicemail emailed to me and notify my cell > phone via SMS is a great way to impress my friends. :-) The inbound > context for my X101P looks someth

Re: [Asterisk-Users] Detect hangup on unanswered POTS call

You could always put a wait in before the answer, ie exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Wait(20) exten => s,3,Answer exten => s,4,Voicemail(u1234) exten => s,5,Hangup HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Asterisk Works on Linux on Sparc

I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I

[Asterisk-Users] Detect hangup on unanswered POTS call

I've been using * at home for a while now and I'm quite happy with how it works. Having voicemail emailed to me and notify my cell phone via SMS is a great way to impress my friends. :-) The inbound context for my X101P looks something like this: exten => s,1,Dial(SIP/analog1&SIP/analog2,20) ex

[Asterisk-Users] ADSI - BT Easicom 1000

Hi Folks, If anyone is interested I now have a BT Easicom 1000 working with *. Some initial problems but they are sorted. Everything seems to work ok with the exception of the vm script. At the moment I basically have to decline the download for it to work.. The really good news is that I guess

Re: [Asterisk-Users] a little oh323 questoin

Dave Alan Caruana wrote: (gdb) bt #0 oss_new (p=0x41be3aa0, state=5) at chan_oss.c:698 #1 0x41bad7d2 in console_dial (fd=1, argc=0, argv=0xbfffe290) at chan_oss.c:902 #2 0x08069b1a in ast_cli_command (fd=1, s=0x41bae940 "Console") at cli.c:1006 #3 0x0807b32e in main (argc=1102817156, argv=0

Re: [Asterisk-Users] a little oh323 questoin

(gdb) bt #0 oss_new (p=0x41be3aa0, state=5) at chan_oss.c:698 #1 0x41bad7d2 in console_dial (fd=1, argc=0, argv=0xbfffe290) at chan_oss.c:902 #2 0x08069b1a in ast_cli_command (fd=1, s=0x41bae940 "Console") at cli.c:1006 #3 0x0807b32e in main (argc=1102817156, argv=0x41be3af4) at asterisk.c:

Re: [Asterisk-Users] a little oh323 questoin

Dave Alan Caruana wrote: doesn't seem to be dumping a core at all if it is, can't find it. Turn it on by running: ulimit -c 100 Michael. Dave - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, June 03, 2003 5:23 PM Subject: R

[Asterisk-Users] Few questions from new user...

Hi there, I've just recently discovered Asterisk via the website and it looks perfect for what I want to do. I've read the handbook draft but still have some questions: I work for a very small company and have almost no budget for hardware. Ideally we'd like to have 3-4 analogue FXO lines comi

Re: [Asterisk-Users] a little oh323 questoin

doesn't seem to be dumping a core at all if it is, can't find it. Dave - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, June 03, 2003 5:23 PM Subject: Re: [Asterisk-Users] a little oh323 questoin > Dave Alan Caruana wrote: > >

Re: [Asterisk-Users] MP3Player

Hi all,   I found that executing the new mpg123 with: mpg123 sample-hold.mp3 sometimes takes a couple of seconds to start playing. Every subsequent command (exactly the same) starts playing immediately. Maybe this causes the timeout in *?   Regards Jan. - Original Message - From

Re: [Asterisk-Users] a little oh323 questoin

Dave Alan Caruana wrote: many thanks Michael, i've modified my extensions.conf ... from a softphone i'm dialling SIP/[EMAIL PROTECTED] which is the address of my asterisk installation. Asterisk quits immediately with a segmentation fault .. -- Executing Dial("SIP/217.168.168.49:5060", "OH323/[EMAI

Re: [Asterisk-Users] E400P

Mark Spencer wrote: You could, of course, contribute the changes needed to make libpri do Q.SIG, and everyone will benefit. :-) What all is required for Q.SIG? Another bunch of Q.931 type messages, basically. It builds on Q.931 to add PBX (i.e. private network) specific features. Regards,

Re: [Asterisk-Users] a little oh323 questoin

many thanks Michael, i've modified my extensions.conf ... from a softphone i'm dialling SIP/[EMAIL PROTECTED] which is the address of my asterisk installation. Asterisk quits immediately with a segmentation fault .. -- Executing Dial("SIP/217.168.168.49:5060", "OH323/[EMAIL PROTECTED]") in new s

Re: [Asterisk-Users] E400P

> You could, of course, contribute the changes needed to make libpri do > Q.SIG, and everyone will benefit. :-) What all is required for Q.SIG? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-use

Re: [Asterisk-Users] a little oh323 questoin

Dave Alan Caruana wrote: hi, just wanted to know what's the proper syntax for an h323 extension. exten => 555,1,dial(SIP/[EMAIL PROTECTED],52,153.207 ) dials SIP extension 723 on IP 216.52.153.207, what is the h323 equivalent of that ?? Using asterisk-oh

Re: [Asterisk-Users] a beginner's SIP question ..

Hi, Dave Alan Caruana wrote: sorry i'm sending so many emails, I always think of something exactly after i've pressed Send .. please be patient with me :) I also have OH323 installed, supposedly correctly, and the same gateway I want to connect to on SIP also supports H323, however i do not know

[Asterisk-Users] Is there a way to play audio to the callee?

Hi, Is there a way to "announce" a call to the callee? For instance, I've answered an incoming call, collected some info and now want to ring an extension, and make an accouncement to that extension before connecting the caller through. Thanks, Steve ___

Re: [Asterisk-Users] a little oh323 questoin

NOTICE[1232188736]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type 'H323' the default channel type created in the startup is OH323, but how do I specify which extension number (723, in this case) it dials to ?? cheers again Dave - Original Message - From: "Erik A

RE: [Asterisk-Users] a little oh323 questoin

exten => 555,1,Dial(H323/216,52,153.207) Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Alan Caruana Sent: Tuesday, June 03, 2003 8:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] a little oh323 questoin hi, just wanted to know what's