Re: [Asterisk-Users] E1/T1 back to back ??

The cross-over cable is what I do between by Asterisk and my Lucent PBX's. Works great!! Peter Svensson wrote: On Mon, 14 Mar 2005, Brett, Gary wrote: Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of

[Asterisk-Users] bashing my head against broadvoice

Folks, I'm having a nightmare trying to get Broadvoice to work. I've followed their howto. I've searched the wiki and archives but still cannot get it to work properly. I can configure * to accept calls from them or make calls to them but not both. This works inbound; register =

Re: [Asterisk-Users] Final call for departments

/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] Codec Selection in Asterisk

-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

[Asterisk-Users] anyone got a 405 to work on a DL380?

Getting nowhere with Digium support. Trying to tell me that their engineers are working on it and that it could be months. Thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] optimumvoice

Being a CableVision customer I get harrassing phone calls from these guys all the time trying to sell me their OV service. Firstly it's closed. They won't allow you to bring anything to their network. Secondly it uses G729 so there's no faxing etc (although you can buy that for extra cost).

Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960

I use it for my 7960's at the house and it works fine. dean collins wrote: I'm not sure if this will work with your cisco's but I can guarantee that it works with the grandstreams. This is what I use to update my 4 phones, running on my main winxp machine and it's free for non commercial use.

Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.

Don't forget to warn your callers about the recording. Tim Mattison wrote: Try the monitor application instead of record. I think that'll do what you're looking for. On Fri, 2005-01-28 at 13:30 -0800, David Shaw wrote: Hello All, I would like to record inbound and outbound calls to and from one

Re: [Asterisk-Users] Asterisk friendly VoIP providers

I use GalaxyVoice and they are fine. No number portability though. Manjit Riat wrote: I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice.

[Asterisk-Users] Cisco 12SP+ firware anyone?

Anyone know where I can lay my hands on some Skinny firmware for some Cisco 12SP+ phone I got at a yard sale this morning? Thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] no caller ID presented from 12SP+

Hi folks, I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor issue with it. When I use it to call another desk I get no CallerID. The receiving phone diplsays asterisk as the CID. Below is the skinny.conf stanza. [2207] device=SEP00308062B006 version=P002L2J2

[Asterisk-Users] iax hardphone

Is there such a beast yet available? I want to equip our exhibition staff with one to plug into their local net feed at whatever exhibition they pitch up at this week. It must be IAX as that's all I allow in through the firewall and it must be a hardphone cos they'll loose an ATA and a hone and

Re: [Asterisk-Users] HP ProLiant server for Asterisk

I have the new card on order via an RMA for my old one. I'll let you know. Mark Peter Childs wrote: Digium support are trailing some new firmware with the TE410P for machines with the Intel E75xx Chipsets that are having issues (such as the DL380 G4). I believe they are confident they have

Re: [Asterisk-Users] How to Create customized audio file to use with ASTCC??

Hi Derek, Yes there is. Take a look at my web pages http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a project to record as many different regional accents (with local lingo) as I could. It started well but as I had to rely on others to create the files (I don't speak with a

Re: [Asterisk-Users] How to Create customized audio file to use with ASTCC??

recording s are a female Irish voice. Derek Mark Phillips wrote: Hi Derek, Yes there is. Take a look at my web pages http://www.g7ltt.com/VoIP/vmfiles.html. You'll see that I started a project to record as many different regional accents (with local lingo) as I could. It started well but as I had

Re: [Asterisk-Users] How to Create customized audio file to use with ASTCC??

That's odd. I don;t controll the rights on the files. That's done by my provider. Lemme check that the files are actually there rather than on my local machine. Mark Daniel Eboa wrote: Same for me with the french file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

Re: [Asterisk-Users] How to Create customized audio file to use with ASTCC??

OK it seems that I can control the rights. I have fixed the problem. The files are available. Mark Mark Phillips wrote: That's odd. I don;t controll the rights on the files. That's done by my provider. Lemme check that the files are actually there rather than on my local machine. Mark Daniel

Re: [Asterisk-Users] Cisco 12SP+ firware anyone?

CallerID and when I call it from my Cisco ATA 286 * crashes!! So far so good however. Mark Jason p wrote: Firmware NO, a good skinny patch for running these phones.. (i have two that work great) http://www.blackratchet.org/chanskinnyplus/ Jason On Sun, 06 Feb 2005 13:24:10 -0500, Mark Phillips

[Asterisk-Users] Broadvoice issues

Hi all, I signed up for BV over the weekend. I have set everything up as per their howto. I can receive calls to my BV number but cannot make any. I'm running CVS head 1/30/2005 so assume that the patch is already in my code. Am I correct? Is it the patch that's stopping me from making calls?

Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

What exactly are you trying to achieve? On my repeater system, I use the RC210 repeater controller with the Phone Patch option. This is then connected to my Cisco ATA and then onto *. Users can initiate phone calls and callers can either command the controller or initiate calls to the radio

Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

Aha, I see where you're going with this. Firstly, why does it have to be SIP? Are you expecting to be able to have users pick up the phone and dial a radio? If not then there are loads of VOIP for radio apps out there. Many run under linux. All use sound card and serial port. Take a look at

Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

Also look for eqso446, 446MHz is what FRS is on the Western Europe and it a UK package. Mark Phillips wrote: Aha, I see where you're going with this. Firstly, why does it have to be SIP? Are you expecting to be able to have users pick up the phone and dial a radio? If not then there are loads

Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

popular. Then there is Echolink, Wires and some others. My plan was to wrte an Asterisk channel driver for each of these Asterisk then could provide inter-system bridging between the various ham VOIP networks, the PSTN and VOIP Telepony. --- Mark Phillips [EMAIL PROTECTED] wrote: Aha, I see where

Re: [Asterisk-Users] FRS *: an actual business use

I see that to be fraught with problems. Speaking as a radio ham, I find that non radio savvy folks have little appreciation of the unique problems surrounding radio users. Sure you'll be able to connect up your phone patch device to a conference (radiophone patchSIP ATA) but your radio users

[Asterisk-Users] anyone using GalaxyVoice?

build of *, Linksys router with ports 5000-5100 1-2 forwarded to the * host. All other things work like FWD, IAXTel and IPTel. Any ideas? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list

[Asterisk-Users] Can't compile Zaptel drivers

-smp-64GB/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/ztdynamic.o Any ideas anyone? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] Troubles with Galaxyvoice

would be greatly appreciated. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] FWD registration faillures

://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Small office requirements - Can this be done?

://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

[Asterisk-Users] Anyone got their Pulver WiSIP phone working with *?

away. If I call the phone by its URI it rings but the calling phone never thinks the call has been answered by the WiSIP. I have 711 as the codec and inband DTMF (which the WiSIP thinks should be 2833). Any ideas ... Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

[Asterisk-Users] problems calling FWD #'s

the time service at FWD. I get connected but I hear nothing. I have firewall holes for the following ports 5060 5082 1-2 Any ideas? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] Anyone got their Pulver WiSIP phone working with *?

PROTECTED] On Behalf Of Mark Phillips Sent: Tuesday, March 16, 2004 3:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Anyone got their Pulver WiSIP phone working with *? Hi Folks, Took delivery of 3 of these today and am having problems. Pulver tech support is pretty much non existant

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

line support, Hold and Transfer functions with this phone via Asterisk. Thanks, Regards, Steven Thomas -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

Re: [Asterisk-Users] Asterix Sip Stack

://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list

[Asterisk-Users] Problems with FWD

=fwd.pulver.com nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw mailbox=3409 My machine is behind a Checkpoint firewall. Its public address 63.88.139.198; private address is 192.168.18.65. All the normal ports are open. 5000-6000 1-2. Ideas? Mark -- Mark Phillips, G7LTT

Re: [Asterisk-Users] Problems with FWD

Message - From: Mark Phillips [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 3:16 PM Subject: [Asterisk-Users] Problems with FWD Hi Folks, Anyone having issues with FWD lateley? It seems that ever since they sent me a notification about my voicemail I've been

[Asterisk-Users] problems with FWD solved maybe?

that the externip thing is starting to work which in turn is why I think this is my FWD problem/solution. Is there a way of defining more than one localnet and localmask? Perhaps something like; localnet=38.249.233.0,192.168.0,0 localmask=255.255.255.0,255.255.255.0 Folks? -- Mark Phillips, G7LTT/KC2ENI

[Asterisk-Users] If you know your party's extension # please dial it now ...

again Whilst writing this I've had a thought. What would happen if I had an entry like this? ; transfer to regular extension # exten = _3XXX,1,Dial(SIP/{EXTN}|20|T) exten = _4XXX,1,Dial(SIP/{EXTN}|20|T) Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

RE: [Asterisk-Users] Galaxy Voice

___ 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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

[Asterisk-Users] CID not appearing via X100P

feed failed: Success Jul 12 16:40:53 WARNING[15374]: chan_zap.c:4980 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Zapateller(Zap/1-1, nocallerid) in new stack Even when there is caller ID on the analogue call. Any ideas? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

Re: [Asterisk-Users] MusicOnHold. not getting it.

__ ___ 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 -- Mark Phillips, G7LTT/KC2ENI

[Asterisk-Users] message waiting notification issues

Grandstream phones and on the Cisco ATA186 but not with either my Pulver WiSIP or X-Ten Pro (yes I did register it and no you can't have a copy) softphones. I'm also having problems with DIAX which uses IAX. Ideas? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

[Asterisk-Users] still got zaptel troubles

-john-not-in-momnt) exten = s,9,Playback(new/theyre-rattlesnake-wrstling) exten = s,10,Voicemail(u${PHONE1VM}) exten = s,11,Hangup exten = s,108,Wait(2) exten = s,109,Voicemail(b${PHONE1VM}) exten = s,110,Hangup Any help etc ... Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http

[Asterisk-Users] can't hear vm audio

work but now it doesn't. Asterisk does show that it is playing the file but no audio is heard. I have audio on regular SIP based calls as well as IAX based ones. I'me not getting and audio when I make a ZAP call. Ideas? Mark G7LTT/KC2ENI Mark Phillips G7LTT/KC2ENI Mark Phillips

Re: [Asterisk-Users] TigerJet ISDN card

Is it CAPI compliant? if so yes Is there any Linux/* support for the TigerJet ISDN card? -brian G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

Re: [Asterisk-Users] can't hear vm audio

that it is playing the file but no audio is heard. I have audio on regular SIP based calls as well as IAX based ones. I'me not getting and audio when I make a ZAP call. Ideas? Mark G7LTT/KC2ENI Mark Phillips G7LTT/KC2ENI Mark Phillips

Re: [Asterisk-Users] Zapata required?

PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

[Asterisk-Users] Zapateller issues

exten = s,3,Dial(a bunch of SIP extensions) But then every call was answered regardless of CID and the tones were heard. Any ideas? G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] Zapateller issues

a chance to sort itself out before the tones play? On Mon, 12 Apr 2004, Mark Phillips wrote: I tried, exten = s,1,Zapateller(answer|nocallerid) exten = s,2,Privacymanager exten = s,3,Dial(a bunch of SIP extensions) But then every call was answered regardless of CID and the tones were heard

[Asterisk-Users] Asterisk, GalaxyVoice and Humble Pie

guy sent you. Enough for now G7LTT/KC2ENI Mark Phillips ___ 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

Re: [Asterisk-Users] FAX?

*** ___ 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 G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users

Re: [Asterisk-Users] VoIP Phone Recommendations

://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

Re: [Asterisk-Users] Fax Over VoIP

://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Dynamic recording function?

up but none that could be done dynamicly. What I'd like to be able to do is press a button and have * start recording the call from that moment and then either stop when I hang up or stop when I press another button. Ideas? G7LTT/KC2ENI Mark Phillips

[Asterisk-Users] weird IAX2 things going on

and C require X100P cards before IAX2 will work correctly? I don't think this is the case because the call can be passed to them after it has been setup via A. Its really bugging me. Any ideas? G7LTT/KC2ENI Mark Phillips ___ Asterisk-Users mailing list

Re: [Asterisk-Users] weird IAX2 things going on

I'm using the guest account that comes in the default setup files and I don't register any of the machines. On Wed, 2004-04-21 at 17:31, Mark Phillips wrote: Hi all, I have 3 * boxes all running the same OS and software version. Machine A has an X100P card, machines B and C do not. They all

[Asterisk-Users] Can asterisk be programmed to make alarm calls?

that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm thinking of something like a alrm call that one has in a hotel room. YOu pick up the phone and program a ring back time. Hope this make sense. Thanks G7LTT/KC2ENI Mark Phillips

Re: [Asterisk-Users] Can asterisk be programmed to make alarmcalls?

de Mark On Thu, 2004-05-13 at 13:41, Mark Phillips wrote: Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which

[Asterisk-Users] Anyone got latest SIP image for Cisco 7960?

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips

Re: [Asterisk-Users] English vs American voice files

files. The few that I've listened to sound fine. How are you getting on with yours Bill? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] English vs American voice files

-- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ 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] English vs American voice files

OK, I've posted the orignal WAV files in 44.1KHZ x 16bit mono format here http://g7ltt.dyndns.org:8010/VoIP/vmukmale-wav.tgz (26MB!) Mark Mark Phillips said: Erm, didn't think of that. Stupidly I deleted the individual wav files. Not a problem though as I have the 3 master files that I

[Asterisk-Users] Galaxy Voice changed their SIP proxy

one call into my system. Ideas? G7LTT/KC2ENI Mark Phillips ___ 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] American vs English

. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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] Galaxy Voice changed their SIP proxy

think folks? On Wed, 2004-09-22 at 17:25, Mark Phillips wrote: I got a call from GV on Monday evening telling me they wanted me to move my Asterisk server over to a new IP address (216.229.127.40) by this saturday. Why the couldn't tell me this in an email is beyond me but anyways .. So I done

RE: [Asterisk-Users] American vs English

Of Mark Phillips Sent: 22 September 2004 23:37 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] American vs English Folks, A few people have made me aware of some omissions in my files (not my fault, they weren't in the Script from the Wiki) which I shall be tackling this weekend. Whilst

RE: [Asterisk-Users] Galaxy Voice changed their SIP proxy

challenge them when I know them to be a peer, secondly if it really must be this way how do I not challenge them? Ideas? Mark On Thu, 2004-09-23 at 13:57, Kevin wrote: I have been using ulaw. -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, September

[Asterisk-Users] SNOM 200 availability

about Snom not shipping the 200 currently to the US but that I could have them in January. Has anyone heard this to be the case? What about other suppliers? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list [EMAIL PROTECTED

Re: [Asterisk-Users] Using my GrandStream remotely

. ___ 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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

[Asterisk-Users] MWI for X-Ten Pro?

Hi Folks, I shelled out for some licences for the X-Ten Pro phone so that we could use it whilst away from the office. Only problem seems to be that I can;t seem to work out how to make it tell me if I have VM without dialing the VM system. Any ideas? Thanks -- Mark Phillips, G7LTT/KC2ENI

RE: [Asterisk-Users] MWI for X-Ten Pro?

: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Monday, October 18, 2004 11:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI for X-Ten Pro? Hi Folks, I

[Asterisk-Users] new quad T1 install

3714? How do I get 3714's CLI to be 732 111 3714 on outbound calls? How does * handle the calls going out? I'm guessing this is something to do with Dial(zap/something)? Any tips etc would be greatly received. Mark -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

Re: [Asterisk-Users] Can I pick up a phone that rings from my phone?

___ 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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ

Re: [Asterisk-Users] Asterisk to Asterisk using SIP?

/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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] tdm410 driver prevents files being played

the zaptel driver for the tdm4xx the lights go out and then any audio file (eg VM) that * wants to play stalls. The CLI says that it is playing the file but nothing is heard and the file never terminates. Hanging up seems to do what is it meant to do. Any ideas? -- Mark Phillips, G7LTT/KC2ENI

[Asterisk-Users] MOH whilst waiting for Conference attendees

Is this possible? I'd like for the members of a conference to hear the hold music until such times as the host of the conference has joined. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] NAT with Linksys

/asterisk-users G7LTT/KC2ENI Mark Phillips ___ 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] Cisco IP phones, SIP, Call-Manager Contracts

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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ Asterisk-Users

Re: [Asterisk-Users] SIP and NAT - best practices?

Most often the simple addition of nat=yes in the relevant sip.conf stanza is all that's required to make a remote SIP phone work from behind a firewall. for example [2201] user=blah secret=blah auth=blah allow=blah host=dynamic nat=yes I've been running 4 remote SIP phones across the

[Asterisk-Users] Anyone interested in getting a basic training course together for the greater NYC area?

Contact me off list if interested. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] dial out and message playback

An example of this would be Outcall Voice Mail? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Danish Samad wrote: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk

Re: [Asterisk-Users] Wifi phone set-up

Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good.

[Asterisk-Users] MOH sourced from a sound card?

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to

Re: [Asterisk-Users] MOH sourced from a sound card?

This looks like the solution. I'll let you know how I get on. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: Hello, MP Can I couple this to the sound card in the Asterisk server and then have MP it play into the MOH? If so how? Yes, it's possible. I've tried

Re: [Asterisk-Users] MOH sourced from a sound card?

=Asterisk+config+musiconhold.conf) and sox with the alsa pseudo-filetype, and output to stdout with the correct bitrate and samples... see the sox manpage for instructions. Untested, but I think that should do the job for you... Mark Phillips wrote: I thought this had been around before but I can't

Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT

Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool box. I can't always access the net whilst on site. If I get stuck doing something I can look it up in the book. Mark, G7LTT/KC2ENI

Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT

It does indeed. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com James Ronald wrote: Does the printed version have an index? -- JR Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool

[Asterisk-Users] change languages from an IVR

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he

Re: [Asterisk-Users] Asterisk native sounds now available!

Kirs et al, I did this already. It's on my website. Your most welcome to use them Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Kristian Kielhofner wrote: Alex Barnes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of

Re: [Asterisk-Users] 911 and ISDN PRI

I dunno about your provider but I know that 2 of my 3 MCI PRI circuits have no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michael Collins wrote: 911 **should** work on a PRI. If you are getting a

Re: [Asterisk-Users] asterisk to FWD

One problem I can see is that you're not using the keys that come with asterisk. Mine (which works!) looks like this iax.conf register = user:[EMAIL PROTECTED] [iaxfwd] type=peer context=from-fwd permit=65.39.205.0/24 auth=rsa host=iax2.fwdnet.net inkeys=freeworlddialup disallow=all

Re: [Asterisk-Users] Asterisk native sounds now available!

Erm ... sorry. That should read Kris et al Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Mark Phillips wrote: Kirs et al, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] Better i18n for Asterisk?

The same 7 sound file is used to indicate both time and quantity. The sound file could be easily recorded to say sept heure but then every time the VM system tells a user that they have 7 messages they'll hear something like vous avez sept heure notification (excuse my schoolboy French).

Re: [Asterisk-Users] change languages from an IVR

I've come across this in my dealings with my customers in Toronto. As an Englishman I find it most infuriating. French is after all, the most hated language in the world from an Englishmans perspective ;-} Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Derek Whitten wrote: Colin

Re: [Asterisk-Users] change languages from an IVR

Aha!! why didn't I think of that. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Gonzalo Servat wrote: On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote: A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get

Re: [Asterisk-Users] Free IAX login

Try adding insecure=very to the guest user account in iax.conf. This should not do a user/pass challenge on the incoming call. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: Not sure answer your question? Try to write some html code and let user register the username

Re: [Asterisk-Users] asterisk to FWD

I forgot to add that you must have an IAX acount with FWD. A regular SIP account won't let you then use IAX. You have to register for it. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Mark Phillips wrote: One problem I can see is that you're not using the keys that come with asterisk

Re: [Asterisk-Users] change languages from an IVR

. -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 07, 2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] change languages from an IVR I've come across this in my dealings with my customers in Toronto

Re: [Asterisk-Users] Asterisk native sounds now available!

to try out. Not a bad price at $79. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Alex Barnes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: 07 February 2006 19:23 To: Asterisk Users Mailing List - Non

Re: [Asterisk-Users] Asterisk native sounds now available!

Is a panoplie legal in Wales? I thought they did away with those at the same time as the Wooly Mountainside Brothels? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Wilson Pickett wrote: I've been looking for someone whom speaks both with a Welsh accent and also the language. Check

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