Re: [Asterisk-Users] IAXTEL toll-free

It's being replaced now with a new Dell. Details will be made available soon. Mark On Wed, 9 Jul 2003, Paul Cheng wrote: Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. www.iaxtel.com

[Asterisk-Users] dbget dbput

Hi, do i need some other software than asterisk to use database commands - dbput and dbget in asterisk ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

[Asterisk-Users] modules.conf again

Hi everybody and sorry for posting this again to the list. I don't want you guys to think that I'm DEMANDING FOR SUPPORT (we all have just had several discusions about that) but my experience tell me that if a posted question is easy enough, it is answered immediatly, or it will never be

[Asterisk-Users] Error on PRI channel : Call specified but not found!

Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup

[Asterisk-Users] ignorepat doesn't work

Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat = 9 exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1} exten = _9[123456789]XXX,2,Congestion this is properly included in the handsets' context but the dial tone

Re: [Asterisk-Users] Transfert call

Yeah ! It is good I will try now Rattana - Original Message - From: carlos del mayor [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 08, 2003 5:22 PM Subject: Re: [Asterisk-Users] Transfert call Sorry! Didn't know it got implemented!!Last notice I had is that it would be

[Asterisk-Users] Error on PRI channel : Call specified but not found!

Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup

[Asterisk-Users] Newbe Questions.

Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks

[Asterisk-Users] Chan_capi hanging channels

Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone got any ideas how to

[Asterisk-Users] error on web page for msn

Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor

Re: [Asterisk-Users] modules.conf again

thanks a lot, E., now I understand it! regards cmayor --- Emanuele Pucciarelli [EMAIL PROTECTED] escribió: Il mer, 2003-07-09 alle 09:31, carlos del mayor ha scritto: THANKS VERY MUCH in advance, and here they are, my two little questions... Well, here are my two little answers, I hope

[Asterisk-Users] Matching winth asterisk-oh323

Hi, It is possible to do matching in oh323.conf with asterisk-oh323? example : alias=0XXX Regards Rattana

Re: [Asterisk-Users] Chan_capi hanging channels

WipeOut . wrote: Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone got

[Asterisk-Users] Asterix Manual

Hi, I am new to Asterix. I would like to know from where I can get a manual on how to use Asterix. Thanks Dhammika

Re: [Asterisk-Users] voip

On Monday 07 July 2003 10:40 pm, [EMAIL PROTECTED] wrote: On Tue, 8 Jul 2003, marrandy wrote: Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx

Re: [Asterisk-Users] Asterix Manual

http://www.digium.com/handbook-draft.pdf Dhammika Gunawardena (ISP) wrote: Hi, I am new to Asterix. I would like to know from where I can get a manual on how to use Asterix. Thanks Dhammika ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Use dialing plan from h.323 gatekeeper?

Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to

[Asterisk-Users] It's true - Nikotel charge for not-completed calls

Hi A few days ago, Kelly remarked that he had previously observed that Nikotel charged him for calls he did not actually complete. I have made a number of test calls to my landline without picking up the calls. I just let it ring once and hung up on the calling phone. A look at the call

Re: [Asterisk-Users] Matching winth asterisk-oh323

Rattana BIV wrote: Hi, It is possible to do matching in oh323.conf with asterisk-oh323? example : alias=0XXX No. In this case you will put in oh323.conf: prefix=0 and then, in extensions, you will do the pattern matching you want. Regards Rattana Michael.

Re: [Asterisk-Users] oh323 prob :)

Dave Alan Caruana wrote: i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait(Zap/1-1, 1) in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial(Zap/1-1, OH323/h323:[EMAIL PROTECTED])

Re: [Asterisk-Users] error on web page for msn

ok, sorry about that folks http://ausfone.com/msn/ should now be used instead ;-) Gary On Wed, 9 Jul 2003 11:29:17 +0200 (CEST), carlos del mayor wrote: Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page

RE: [Asterisk-Users] Use dialing plan from h.323 gatekeeper?

Hi, I think I understood how to achieve this. Anyway, a working config is welcome if anyone has already done it. hristo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HT Sent: Wednesday, July 09, 2003 2:54 PM To: [EMAIL PROTECTED] Subject:

[Asterisk-Users] PBX / Asterisk integration

Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX

Re: [Asterisk-Users] ignorepat doesn't work

Hi bk, On Wed, Jul 09, 2003 at 17:16:55 +0900, BK [address only for mailing lists] wrote: Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat = 9 exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1} exten =

Re: [Asterisk-Users] error on web page for msn

woops... http://phone.nq.net/msn might actually find it... it wil be moving soon to http://www.ausfone.com/msn after some reorganisation here... On Wed, 9 Jul 2003 11:29:17 +0200 (CEST), carlos del mayor wrote: Hi everybody, I'm trying to use msn with * and for that, I'm reading all

RE: [Asterisk-Users] Net2Phone SIP

Just to say that I've now managed to get this going by pretending to be an ATA 186. change the User-Agent string in chan_sip.c from Asterisk to Cisco ATA 186 and the Net2Phone Sip service works with * Would it be possible to pick this up from sip.conf in a future release? Regards Mark

Re: [Asterisk-Users] modules.conf again

1)As I have seen, to make Asterisk load chan_capi.so and chan_modem.so you must have: load=chan_capi.so and load = chan_modem.so in your modules.conf. But I had understood some time ago that setting autoload = yes made Asterisk load every module that was necesary. Then, why must I load these

Re: [Asterisk-Users] Error on PRI channel : Call specified but notfound!

That's a new one, but contact Martin and he should be able to help assist you. Mark On Wed, 9 Jul 2003, Cristi wrote: Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File

Re: [Asterisk-Users] Chan_capi hanging channels

WipeOut . napsal(a): Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone

Re: [Asterisk-Users] PBX / Asterisk integration

I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so

[Asterisk-Users] Music on hold quality..

Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of

[Asterisk-Users] more abou msn

Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___ Yahoo!

[Asterisk-Users] caller id

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971

Re: [Asterisk-Users] Music on hold quality..

On Wed, 2003-07-09 at 08:57, WipeOut . wrote: Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all

Re: [Asterisk-Users] ignorepat doesn't work

thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because

[Asterisk-Users] callerid= being ignored

Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= Full name 1001 etc etc Now, when I do this in a given extension exten = ,1,NoOp(${CALLERIDNUM}) then I get gs1 as callerid and not 1001 as defined with callerid= Sure, I could set the usernames to their

Re: [Asterisk-Users] Music on hold quality..

No I don't have any Zap channels, I am using chan_capi for my PSTN connection and SIP only.. Does moh depend on zap?? On Wed, 2003-07-09 at 08:57, WipeOut . wrote: Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes

[Asterisk-Users] incoming callerid on FXO

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the

[Asterisk-Users] chan_h323, Asterisk and DTMF issue

Hi folks, I’m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won’t work with Asterisk’s voicemail system. I’m using the

RE: [Asterisk-Users] Accurate Billing

My friend has a small pay phone services business running using analog lines and a conventional PBX. He allows a delay before starting billing. So if a customer's call is not answered after the allowed delay he is billed. Shepherd

RE: [Asterisk-Users] Music on hold quality..

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: 09 July 2003 16:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold quality.. No I don't have any Zap channels, I am using chan_capi for my PSTN connection and SIP only..

Re: [Asterisk-Users] Music on hold quality..

On Wed, 2003-07-09 at 10:12, WipeOut . wrote: No I don't have any Zap channels, I am using chan_capi for my PSTN connection and SIP only.. Does moh depend on zap?? Yes, it uses Zap for timing. Look into ztdummy, or the rtc driver. Is there any documentation on using the ztdummy or rtc

[Asterisk-Users] SUMMARY: Problems with Hangup Detection in VoiceMail2.

Many thanks to Martin Pycko and Mark Spencer. Mark's suggestion below was correct: Maybe it's stuck trying to send the e-mail notification. If you take the e-mail address out of /etc/asterisk/voicemail.conf does that speed it up? Indeed it did! The problem turned out to be a 60second delay

Re: [Asterisk-Users] caller id

Use SetCallerID(1234567). Tan telappliant.com - Original Message - From: Marian Danisek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 09, 2003 3:23 PM Subject: [Asterisk-Users] caller id Hello, is it possible to change how are caller id on incoming call from isdn,

Re: [Asterisk-Users] MGCP-H323v2 transcoder?

Sure RTFM Jeremy McNamara Sebastian Sill wrote: Hello, I have some MGCP VoIP gateways and some H323v2 VoIP gateways, Can a use the Asterisk for interconnect the VoIP boxes? If I can anyone knows how to configure it? Thank you very much Best regards Sebastian Sill. Uruguay.

Re: [Asterisk-Users] callerid= being ignored

At the moment asterisk can get the callerid from the From: field. regards Martin On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote: Hi I have defined my SIP phones like this ... [Sip1] username=gs1 callerid= Full name 1001 etc etc Now, when I do this in a given

[Asterisk-Users] E1-RJ45 pin configuration

hi! We have ISDN/PRI E1 lines which needs to be connected to the E400P card . Can somebody help us with the PIN configuration of RJ45 in relation with the E1(ISDN/PRI) ? urs, DenZel. ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] PRI with variable length numbers

Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is

Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls

Guys, unless their site states something to the contrary you don't have a hope in hell with this. you are paying them for a voip circuit, which you are using to attempt a call. you have taken up several seconds of voip bandwidth which they are charging you for, the same way you would pay if

[Asterisk-Users] experience with multi-port SIP/FXS gateways?

I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack

Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls

On Thursday, July 10, 2003, at 02:59 AM, Jon Pounder wrote: Guys, unless their site states something to the contrary you don't have a hope in hell with this. you are paying them for a voip circuit, which you are using to attempt a call. you have taken up several seconds of voip bandwidth

Re: [Asterisk-Users] PRI with variable length numbers

Hey Martin, I'm not receiving fixed-length numbers on that PRI and it really seems to be the Asterisk end which decides when dialling is complete. I've arranged for a block of numbers, starting with 3, to be routed from the Meridian to Asterisk, over this PRI. As long as the numbers I set up in

Re: [Asterisk-Users] asterisk-oh323 v0.5.3

Hi Michael, are you adding ilbc support to your channel ? On Tuesday 08 Jul 2003 12:07 pm, Michael Manousos wrote: Hello all, I have updated the asterisk-oh323 package. The new version has several improvements (fixes in audio/RTP stream generation, music-on-hold working, flash hook

[Asterisk-Users] OpenBSD version???

John Todd's Onlamp article mentions an OpenBSD version as of June 2003. Have I been sleeping while reading asterisk-users? Is it a seperate project or is it just making Asterisk portable? Who is working on this and is it in the main CVS yet? Do they have device drivers ported or just the

[Asterisk-Users] Asterisk Call Manager doc

I am looking for a doc out there that on how to use the Asterisk Call Manager. Can someone let me know what the URL to this is. Thanks, John Haigh ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] H450 problems

Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is

[Asterisk-Users] How to modify dialed number?

Hi! Is there simple way how to add prefix to dialed number? I need change 0X. to 0X. Regards Petr Michálek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] IAX2 Warning

When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port for now == Registered

Re: [Asterisk-Users] How to modify dialed number?

exten = _0X,1,Dial,Zap/g1/0${EXTEN:1} Martin On Wed, 9 Jul 2003, Petr Michálek wrote: Hi! Is there simple way how to add prefix to dialed number? I need change 0X. to 0X. Regards Petr Michálek ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] Budgetone and Voicemail

I have a Budgetone 102 with the latest firmware 1.0.3.72 and using dtmfmode=rfc2833 With g711 I have no problem with Voicemail or Voicemail2. With g729 it always repeats digits and it is impossible to check my voicemail (or any other apps that require digits) - Original Message - From:

Re: [Asterisk-Users] How to modify dialed number?

On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote: Is there simple way how to add prefix to dialed number? I need change 0X. to 0X. how about this exten = 0X.,1,Dial(0{EXTEN:1}) rgds bk ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] IAX G729 Codec

Hi, I have recently purchased some Asterisk G729 Codecs and installed them to overcome by bandwidth problem I was having with GSM. The G729 keeps the pings nice and low, but the audio stutters or jitters a fair bit. (Starts and stops) Any Idea what would be causing this ? I am just

Re: [Asterisk-Users] How to modify dialed number?

You forgot about _ in front of 0X Martin On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote: On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote: Is there simple way how to add prefix to dialed number? I need change 0X. to 0X. how about this exten =

Re: [Asterisk-Users] IAX2 Warning

IAX2 uses hardcoded 4569 port so it's not looking for port keyword. Nothing to worry about. Martin On Thu, 10 Jul 2003, Richard Scobie wrote: When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action

Re: [Asterisk-Users] Budgetone and Voicemail

Yes! It did work with g729 and dtmfmode=info. Thanks a lot! - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 09, 2003 7:35 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail try dtmfmode=info solved all my former problems

Re: [Asterisk-Users] IAX G729 Codec

On Wed, 2003-07-09 at 18:06, Jay Tyndall wrote: Hi, I have recently purchased some Asterisk G729 Codecs and installed them to overcome by bandwidth problem I was having with GSM. The G729 keeps the pings nice and low, but the audio stutters or jitters a fair bit. (Starts and stops)

Re: [Asterisk-Users] H450 problems

Hello Aaron Martin, Yesterday we discussed the possibility to buy some Micronet hardware. We haven't used hardware from this vendor yet and I wanted to ask you a question about the Micronet SP5052 FXO Gateway features. Is there any way to setup SP5052 FXO to answer a call from PSTN and to

Re: [Asterisk-Users] H450 problems

Yes this is exactly what I am doing. A caller dials in on the PSTN, and gets connected to the Asterisk server, which answers with Please dial the extension you require. The caller then dials 8000 and gets transferred to extension 8000. However, I cant seem to get transfer working, ie if they

RE: [Asterisk-Users] Asterix Manual

thanks -Original Message- From: carlos del mayor [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 09, 2003 7:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterix Manual Here you can find some documentation: http://www.digium.com/index.php?menu=documentation Regards cmayor

Re: [Asterisk-Users] Billsec on CDR

Steve Can you please give us some guidance on how to make call progress work outside the US or UK? Thanks Dan - Original Message - From: Stephen Davies [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 21, 2003 4:51 AM Subject: Re: [Asterisk-Users] Billsec on CDR On Fri,

Re: [Asterisk-Users] IAX G729 Codec

This is going across a 256k/64 to a 512k/128. They are about 2 hops away from each other and ping times are sub 70ms. (Even when the * audio is playing) - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 10, 2003 12:02 PM Subject:

Re: [Asterisk-Users] IAX G729 Codec

On Wed, 2003-07-09 at 23:30, Jay Tyndall wrote: This is going across a 256k/64 to a 512k/128. They are about 2 hops away from each other and ping times are sub 70ms. (Even when the * audio is playing) What kind of slow as hell routers are you using? The ping times from my office to home is in