[Asterisk-Users] snom 220 busy all the time

Good day all We have a snom 220 that for some reason keeps on giving this message Got SIP response 486 Busy Here back from 192.168.21.222 even though there is no active calls to it and there are 2 accounts set on the phone? Please Help and advice Thanks Altus

Re: [Asterisk-Users] Using Codec G-726

had the same thin with 729 I had to go disallow=all allow=g279 On Thu, 2005-03-17 at 16:37, Matt wrote: Hi, What do I need to do to get Asterisk to allow me to use codec G-726? I've already tried allow=all in my sip.conf config.. didn't work... ___

[Asterisk-Users] snom 220 version

Good day all What is a good stable snom 220 firmware version. Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Fax and Voice

google asterisk fax On Thu, 2005-03-24 at 11:53, Guy Decarpentrie wrote: Hi all, Is * able to do the difference between Fax and voice, and then adapt the treatment of the call ? An example ? Thx ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Fax and Voice

exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit : google asterisk fax Well, i know how

Re: [Asterisk-Users] Fax and Voice

sorry exten = fax,1,Dail On Thu, 2005-03-24 at 12:53, Altus Snyman wrote: exten,fax,1,Dail( On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit : On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars

[Asterisk-Users] snom220 problem

Good day all I have a snom 220 with the extra keypad When more than one call comes in none of the extra lines on the phone lights up or anything.You hear the beep in you ear but no way of picking it up.I tied 4 different firmware versions.On was a very old one,with actually worked but is gave echo

[Asterisk-Users] sox

Good day all I previously tried the Monitor app with sox but it did not work and according to the list it was because of a broken version What are a good and working version for the latest asterisk Thanks altus ___ Asterisk-Users mailing list

[Asterisk-Users] snom220

Good day all I'm looking for someone with good knowledge of the way the snom220 transfer I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to 200 1st and the transfer B

RE: [Asterisk-Users] snom220

Does Call join on Xfer (2 calls) be on or off? Thanks On Fri, 2005-04-01 at 04:29, Damon Estep wrote: I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to

[Asterisk-Users] Planet VIP 450

Good day all Did someone get the planet VIP 450 working Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] fedora 3

Thanks for the trouble n Wed, 2005-04-06 at 15:00, iMRAN wrote: Hi, I`ve installed on FC-3 last month and its working gr8... no probs so far Imran On Apr 6, 2005 2:38 PM, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I have a Fedora core 3 installation Is there any

[Asterisk-Users] voicetronix dtmf

Good day all I got the latest cvs asterisk But when making a call out threw the voicetronix openline4 card the dtmf doens not work I got this in vpb.conf ecsuppthres = 4096 indication = 1 dtmfidd = 3000 ast-dtmf-det=1 relaxdtmf=1 break-for-dtmf=yes Please help Thanks Altus

[Asterisk-Users] pbx to asterisk

Good day all I just want to know if someone tried this and with out any hassles What I want to do is take 4 extension(analog) of a current,old,pabx unit and put them into a asterisk server with a 4port analog card,like the voicetronix openline4 card. (PSTN)(old PABX)---===(4 ports

[Asterisk-Users] voicetronix bri

Good day all Will a voicetronix openline 4 card work with a 4port BRI card? Please HElp/advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] voicetronix bri

Voicetronix will only be used for the gsm cell router and BRI for outgoing-incoming calls On Thu, 2005-04-14 at 11:26, Michael Bielicki wrote: In what sense ? voicetronix is analog BRI is ISDN digital On 4/14/05, Altus Snyman [EMAIL PROTECTED] wrote: Good day all Will a voicetronix

[Asterisk-Users] qos test

Good day all I'm looking for a type of QOS test tool(software) I want to test if a link is good enough for voip and test witch ones will be the best..ens any ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Call(out) routing

Good day all I had a look at the extensions.conf sorting http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting What I'm trying to do is route all my cellphone number threw a channel and all other calls threw the other 3 channels Cellphone numbers are 10 number,i.o.w XX.

[Asterisk-Users] fax to email

Good day all I have a pri card,e100 What I want to do is If a fax comes in for number 1234567890 it should be e-mail to [EMAIL PROTECTED] If a fax comes in for number 0987654321 it should be e-mail to [EMAIL PROTECTED] ens Can this be done and how

Re: [Asterisk-Users] fax to email

and email-fax?? The other way around On Thu, 2005-01-06 at 14:17, Andrew Kohlsmith wrote: On January 6, 2005 06:55 am, Altus Snyman wrote: Good day all I have a pri card,e100 What I want to do is If a fax comes in for number 1234567890 it should be e-mail to [EMAIL PROTECTED

Re: [Asterisk-Users] fax to email

How do I fax a .tiff file with asterisk? On Thu, 2005-01-06 at 15:13, Michael Welter wrote: Altus Snyman wrote: and email-fax?? The other way around You can run a simple mail server on the * box to accept emails addressed to the .fax domain (i.e. [EMAIL PROTECTED]). This presumes

[Asterisk-Users] fax e-mail spandsp

I'm trying to install spandsp But when I try to patch the Makefile it gives this error [EMAIL PROTECTED] apps]# patch apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is

Re: [Asterisk-Users] fax e-mail spandsp

the changes in the apps/Makefile have progressed while the patch makefile have not. Here is a current patch that works as of CVS-HEAD-01/06/05-14:47:06 Regards, Jim On Fri, 7 Jan 2005, Altus Snyman wrote: I'm trying to install spandsp But when I try to patch the Makefile it gives

[Asterisk-Users] TE110P error

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10

RE: [Asterisk-Users] TE110P error

? A better guess is either the driver for the card isn't loaded or the zap config files aren't agreeing with each other. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: Monday, January 10, 2005 1:24 AM To: asterisk Subject

[Asterisk-Users] error?

Good day all I'm getting this error out of the blue on a incoming call? Any idea?Pleas Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ILBC since our native format has changed to SLINR ___ Asterisk-Users mailing list

Re: [Asterisk-Users] fax e-mail spandsp

Did anyone get asterisk to actually work with a fax coming in on a pri number and e-mail it to a user? On Mon, 2005-01-10 at 08:29, Howard Lowndes wrote: On Mon, 2005-01-10 at 16:00, Altus Snyman wrote: Its still fails! [EMAIL PROTECTED] apps]# patch apps_makefile.patch.new patching

[Asterisk-Users] snom220

Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can

Re: [Asterisk-Users] snom220

Sorry It works Just had to reboot the phone On Thu, 2005-01-13 at 08:40, Altus Snyman wrote: Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added exten = 403,hint,SIP/403 in my dialplan But These lights only comes

[Asterisk-Users] Grandstream bt-100 loosing it!

Good day all We have one Bt-100 that logs on to the server,works for a few min and then just starts loosing registration Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145' Jan 13 13:10:05 NOTICE[-1101505616]:

[Asterisk-Users] sip-sip

Good day all We have a asterisk server running sip for about 20 users We have a client running a unknown sip server in a different country I phone the guy there and he gave a a account(username+password) What I want is if a users calls the number of that country it should be send to the sip server

[Asterisk-Users] ilbc high bandwidth

Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well Is there a way to bring it down? Pleas Help Altus ___ Asterisk-Users mailing list

[Asterisk-Users] h323

Good day all I have a asterisk server running sip and sip phone How do I get asterisk to call another h323 server? Please Help Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] h323 client

Good day all Just to re phrase my previous question We have asterisk running sip for sip phone In the US there is a h323 server What I want to do is: All calls coming into my pbx via sip thats got a american number to go threw the h323 server I have set this up with 2 sip servers where the one

[Asterisk-Users] Grandstreams+Nat

Good day all I cant get my grandstream bt-100 to register My asterisk is on a public ip and the phone behind a nat firewall I added nat=yes in sip.conf and did this on my grandstream set the GS to SIP server=asterisk.yourhost.com and leave Outbound Proxy empty * set the GS to SIP port 5060 and

[Asterisk-Users] Dialplane slip

Good day all My extensions.conf is something like this [main] ;---incoming+ play welcome message extens = s.. ;---users extensions exten = 100. ;---outgoing ignore 0 ;- It all works fine The message says dial 1 for this ens But if I dial 0+number it will actually make

[Asterisk-Users] asterisk remote monitor

Good day all We have a few remote pbx systems running I would like to monitor the and check that they are up and running and working Please Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] BRI only 2 calls

Good day all I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3 This is to install my quad bri card All installed well I coped over some old config files.All 4 ports are available,so that gives 8 open lines for incoming or outgoing,correct me of I'm rond The problem is,asterisk

[Asterisk-Users] why asterisk and ser

Good day all Why would u use asterisk and ser together and what is the big difference? Thanks altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] warning message

Good day all.I get the warning message on my system,this is for a snom 220,it repeats this message a few times,please help Feb 8 09:29:26 WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Non-critical Request) Is there a page that

Re: [Asterisk-Users] snom soft phone

Did you try 00 That is what it is on the 220 On Tue, 2005-02-08 at 09:36, Paradise Dove wrote: what is the password for Administrator in the softphone? On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke [EMAIL PROTECTED] wrote: Go to the web page, in Preferences there are two

Re: [Asterisk-Users] How to xfer calls or is my setup wrong?

What asterisk version I know we had a problem with one of the cvs We couldn't use the transfer buttons,but # worked What about the Dail(SIP/111,12,tT) in your extensions.conf On Tue, 2005-02-08 at 13:50, Mark Benson wrote: I am having problems transferring calls from one sip extension to

[Asterisk-Users] spandsp

Good day all I have a asterisk installation,1.0.3, and spandsp. I got asterisk working,I edited the make file myself. Now when I receive a fax I only get half a page or nothing any Ideas why Please let me know Altus ___ Asterisk-Users mailing list

[Asterisk-Users] bri dropping calls

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

Re: [Asterisk-Users] bri dropping calls

O did not have a look at it yet,I got the one from a week ago,how is aterisk 1.0.5? On Wed, 2005-02-09 at 08:04, Michael Bielicki wrote: hmmm the latest bristuff uses asterisk 1.0.5 so it can't be laast, can it ? cheers Michael On Wed, 09 Feb 2005 07:24:34 +0200, Altus Snyman [EMAIL

[Asterisk-Users] sip_notify.conf

Good day all What is the file sip_notify.conf for Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] bri dropping calls

Where do you get this new version of bristuff,I had a look on the webpage and there's only RC3 On Wed, 2005-02-09 at 08:58, Peer Oliver Schmidt wrote: Altus Snyman wrote: We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3

[Asterisk-Users] limit iax calls

Good day all We have 2 asterisk servers,connected with iax2 and the phone via SIP They dont have a very big line so I want to restrict the call limet to 3 iax2 calls at a time,and for instance it the 4th call is made it will say something like all lines are being use try later Please help thanks

Re: [Asterisk-Users] Cisco7960/SCCP Transfer Help?

If you select more there Trnsfer and BlndXfer will be displayed BlndXfer for Blind transfer Trnsfer for Confirm transfer This is on 7960 On Thu, 2005-02-10 at 15:09, [EMAIL PROTECTED] wrote: I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest

[Asterisk-Users] Bri problem

Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension s?? Is this something to do

Re: [Asterisk-Users] Bri problem

Thanks Will have a look On Fri, 2005-02-11 at 09:59, Edin Kozo wrote: Hi Do you have immediate=no in your zapata.conf ? immediate = yes makes asterisk pass all incoming calls to s extension. Hope that helps you --- Altus Snyman [EMAIL PROTECTED] escribió: Good day all I've installed

[Asterisk-Users] spandsp asterisk 3/5

Good day all I want to know with version of spandsp works well with ether asterisk 1.0.3 or 1.0.5 Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] asterisk in New-Zealand

Good day all Anyone doing asterisk in New-Zealand? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Asterisk in Singapore.

I can get you a good deal if you import the from South-Africa..Let me know.Altus On Mon, 2005-02-14 at 15:38, Jonathan Gill wrote: In the vain of asterisk in new-zealand... Anyone know of a reliable source of digium gear in singapore? Also where to pick up IP phones, anyone any clues? Ta

[Asterisk-Users] asterisk qualified

Good day all Is there any time of VOIP/SIP/asterisk qualifications or certificates? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] h323

Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] Sangoma A104 - D-Channel problem

While on sangoma We are getting a samngom pri?Is there any driver I need to install,how does it work,like a Zaptel card. Any doc Please Let me know altus On Fri, 2005-02-18 at 11:06, Kumak wrote: On Fri, Feb 18, 2005 at 03:38:28AM +0100, Michael Bielicki wrote: upgrade to the following

[Asterisk-Users] Sangoma A101

Good day all Is there any difference in the sangoma zaptel.conf and zapata.conf then other cards Thanks altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] route outgoing call

Good day all I registered at a few sip server in different countries Now I want to route outgoing calls for that country threw that sip server and all the others there my own pstn,ZAP card.I already registered asterisk with them. How would my extensions.conf look.This is what I have but no matter

Re: [Asterisk-Users] does asterisk support menus?

Yes Application Background() On Tue, 2005-02-22 at 14:35, Muhammad Muzzamil Luqman wrote: Whenever some call comes in i want it to be automatically picked up and then it plays some message Welcome to xyz, Press 1 for sales and 2 for support and then it takes it to the particular extension of

[Asterisk-Users] send fax with pri

HI all What is the best to send a fax with a PRO. I got it working on the receiving and e-mailing it.How do I send one Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] hylafax

Good day all Can hylafax work with asterisk..and how I'm trying to find a way to send a fax over my E1 connection Please Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Difference between E1 and PRI

PRI comes in 2versions E1 European and T1 US E1 30 channels T1 23 channels On Wed, 2005-02-23 at 14:15, Eric Bishop wrote: Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the

[Asterisk-Users] snom220 *8 hangup

Good day all We have a snom 220 set as a switchboard phone I also configured *8 so that if the operator is somewhere else and it rings she can just go *8 on the nearest phone,Grandstrams bt-100 and snom 190.But If she does this she only speaks for about 30s and it will cut off the caller? Any

[Asterisk-Users] IAX+G729a

Good day We are going to add 6 channels of G729a to our asterisk server running iax between them I have a few question about the hole license thing. In iax.conf do i allow g729 or g729a?What's the difference? This license is for 2 servers,i.o.w 3 per server.How many calls does this give us? For

[Asterisk-Users] iax,trunking,zap

Good day all Why do I need a Zaptel card to do trunking in IAX?? What if I only had a voice/iax router? Is there a way around this? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] from sip to asterisk to h323..how

Goo day all This is our setup Client phone--(SIP)--asterisk server---SIP/IAX---asterisk--- -- goes out to international server running sip/iax But now I want to dial out to H323 server? I.O.W I want asterisk to act as a H323 client that will rout some calls out to a H323 server.How do I do this

[Asterisk-Users] no hangup

Good day all. I have a small problem When someone calls in from the outside,dial the extension of the internal sip phone,and the hangs up without getting any response,the sip phone will keep on ringing? It show that is hangs up on the Zap/vpb channel but the connection between asterisk and sip

[Asterisk-Users] new version problem

Good day all I upgrade my asterisk and the vpb driver to the latest I used all my previous, working, config files over. Every thing works well but for 1 thing,playing DTMF when making a outbound call If I call a external number on my phone and another pbx answers and I have to press a number it

Re: [Asterisk-Users] Problem with sox

I installed the new version of asterisk But the other probelm I got was,were are using the voicetronix cards,so if you go and put ignorepat = 0 exten = _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten = _0.,2,Monitor(wav,${CALLFILENAME},m) exten = _0.,3,Dial(vpb/1-3/${EXTEN:1}) exten =

[Asterisk-Users] hangup()???

Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the

[Asterisk-Users] dail cli

Good day all How do I dial from the cli It says dial [number] but that doesnt do anything? Thanks altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Forwarding calls

What phones are you using I know the grandstreams you have *70somethings in,bin a while The phones have these settings,Mitel 5055 you can do it on the phone web pages ismaelg wrote: Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on

Re: [Asterisk-Users] Grandstream Firmware 1.0.5.16 Attended Transfer

There is a version 18! Michael Nolan wrote: On Wed, 24 Nov 2004 11:02:36 +, Bob Goddard [EMAIL PROTECTED] wrote: On Wednesday 24 November 2004 10:39, Simon wrote: I've searched for a few days now without finding an answer. The release notes for

[Asterisk-Users] No hangup(vpb)

Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] No hangup(vpb)

How and what and where? Sorry I'm a bit new to asterisk and programming Thanks Altus el Flynn wrote: Altus Snyman wrote: Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why

Re: [Asterisk-Users] grandstream bt100

Do you have the callerid thing in sip.conf? And the setting on the phone "User ID is phone number:" = no Myne works Rodney Acosta Coya wrote: hi all, i have some grandstream bt100 registered with asterisk when a extension receive a call display its own number but i need to diplay the

Re: [Asterisk-Users] Receptionist Phone

What about the mitel 5220's buttens? Tracy R Reed wrote: On Sat, Dec 04, 2004 at 08:03:03PM -0400, Cian O'Sullivan spake thusly: She is an older lady and does not want to use a web interface. Any suggestions? Give her a Snom or Polycom phone which does have this

[Asterisk-Users] new version problems

Good day all We got the cvs yesterday,and it seems as if transfer does not work.We are using mitel 52205055 and the Grandstream bt-100,using the transfer buttons. If you transfer it just goes to the next step? please Help Thanks Altus ___

[Asterisk-Users] sangoma

Good day all Is there someone that's got asterisk working well with a A101/E1 card Apparently they don't have RBS support? Please advice Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] sangoma

I want to use it in my pbx as a bri card for incoming and outgoing calls in asterisk How did you get it working with asterisk,drivers ens. Please Help Andrew Kohlsmith wrote: On December 8, 2004 02:40 am, Altus Snyman wrote: Is there someone that's got asterisk working well

[Asterisk-Users] conference call

Good day all We have a Mitel 3300 connected to a grandstream handytone 486 These is a conference unit,one big speaker phone,my question is how to make a conference call using asterisk Other phone has the conference button on so if you press it you can call someone else and all can talk together

Re: [Asterisk-Users] conference call

Sorry its a Mitel 5305 Altus Snyman wrote: Good day all We have a Mitel 3300 connected to a grandstream handytone 486 These is a conference unit,one big speaker phone,my question is how to make a conference call using asterisk Other phone has the conference button on so if you press it you can

[Asterisk-Users] BT-100 Transfer!!

Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to

RE: [Asterisk-Users] BT-100 Transfer!!

- From: Altus Snyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, December 09, 2004 4:21 PM Subject: [Asterisk-Users] BT-100 Transfer!! Good day all We have Grand Stream BT-100 phones The transfer button work well

RE: [Asterisk-Users] BT-100 Transfer!!

We got version Asterisk CVS-HEAD-09/01/04-11:36:41 and all works well for us with the voicetronix card Then we upgrade to a newer version of * and it did not seem to carry over the DTMF signal out of the PSTN Voicetronix fixed this so we got the new version of asterisk and it does not allow us to

Re: [Asterisk-Users] BT-100 Transfer!!

it is a good idea when users get the cvs version.In this way when there are more a chance of someone discovering a bug/problem. On Fri, 2004-12-10 at 17:39, Bob Goddard wrote: On Friday 10 December 2004 14:47, Altus Snyman wrote: Please do not top post. We got version Asterisk CVS-HEAD-09/01/04

[Asterisk-Users] openline4

Good day Does Asterisk work with the Voicetronix Openline4 cards? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] openline4

The thing is,Im not the sharpest tool in the shed, and I really need help setting it up.I've installed Asterisk but thats how far I'm getting,would you please Help me,Please On Mon, 2004-03-29 at 14:33, michiel betel wrote: Altus Snyman wrote: Good day Does Asterisk work with the Voicetronix

[Asterisk-Users] hardware/software needed

Good day all Now I want to install a complete pbx system on my linux box with windows clients. Now I have the a openline4 card and 4 lines,but what software do I need,Asterisk running on the server and?and what for the clients,I see in the config there is a sip provider config?? Thanks Altus

Re: [Asterisk-Users] openline4

Did you get it working,its been 7day and 7 nights and I cant dial out?It receives the demo call but thats it.Please help me On Mon, 2004-03-29 at 14:33, michiel betel wrote: Altus Snyman wrote: Good day Does Asterisk work with the Voicetronix Openline4 cards? Yes, see: http

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[Asterisk-Users] sip no sound?

Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call

[Asterisk-Users] SIP soft?

Good day When I call in from the outside(PSTN),my box answers with the demo,but it is very soft,and when I dial the extension to my client the connection is very soft as well,Please help Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] GUI?

Good day all I'm looking for a GUI/Web interface for Asterisk. What I need it for is to see who's line(SIP) is busy work? Something like a switch board? Please give me some info? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

please let me know if anyone get this..please On Thu, 2004-04-08 at 13:21, Joe Dennick wrote: I'm still having problems being able to get the Transfer function to work. I enter the correct password, but still can transfer or end calls with the Flash Panel. Any suggestions? Joe

Re: [Asterisk-Users] Web interface for Asterisk

ok this is what I did I moved all to my /var/www/html/control. did the changes is my files and used the copy of manager.conf. I started asterisk and did /var/www/html/control/op_server.pl and pointed my browser to 192.168.0.1/control/html ... had the same problem. Then I went and set debug to 1

[Asterisk-Users] transfer sip

Good day all. I need a windows client that can transfer calls from 1 user 2 another with a nice GUI for non PC iterated people Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] Web interface for Asterisk

http://sip.house.com.ar/operator/ On Thu, 2004-04-08 at 16:01, Steve Foy wrote: Hi again :) Can you give me a URL for the software you mentioned? Cheers, Steve On Thu, Apr 08, 2004 at 09:45:47AM -0400, Jain, Sonal wrote: I installed the flash operator panel and I also installed the

[Asterisk-Users] sip client

Good day. I'm looking for a sip client 4 fedora??? Frdora? Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

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