[Asterisk-Users] D-Channel on span 1 down

2004-03-31 Thread zouhair echchelh
Hi, Any Idea about this message error are welcome, this cause communication cuts. == D-Channel on span 1 down MFE for TEI = 84 == D-Channel on span 1 up this is my zapata.conf file ;signalling = bri_cpe pridialplan = unknown ;echocancel = yes echocancel = no immediate=no ; BRI line plugged into

[Asterisk-Users] Anyone has a working * with E1 in Mexico E1 R2 modified?

2004-03-31 Thread Otto Krumm
    I was wondering if anyone has setup an * connected to E1 in Mexico?, what card would you recomend and do you have some info, examples or everythig else... or for instance this setup works?       Thanks in advance       Greetings Otto Krumm  

[Asterisk-Users] D-Channel on span 1 down

2004-03-31 Thread zouhair echchelh
Hi, Any Idea about this message error are welcome, this cause communication cuts. == D-Channel on span 1 down MFE for TEI = 84 == D-Channel on span 1 up Best Regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/l

Re: [Asterisk-Users] safe_asterisk with non-root user

2004-03-31 Thread Ryan Courtnage
Turned out to be a dumb mistake on my part. when su'ing to my non-root user, I was losing my PATH info. safe_asterisk assumes /usr/sbin is in the PATH. Cheers Ryan On 31-Mar-04, at 5:59 PM, Ryan Courtnage wrote: Hello, I've found a couple of previous posts on this subject, but with no posted

Re: [Asterisk-Users] Virbiage Phones - Vapourware??

2004-03-31 Thread Adam Hart
Aaron Martin wrote: Has anyone heard any more info about the Virbiage FT201 VoIP phones? About 3 months ago I was told they were 6 weeks away, about 3 weeks ago I was told they were 2 weeks away, and now I am told they are 2 months away again! Are they EVER going to arrive? Can anyone shed

Re: [Asterisk-Users] Can't compile asterisk.

2004-03-31 Thread Masakazu Nakano
(BIs this same result? (B (Bhttp://lists.digium.com/pipermail/asterisk-users/2003-November/027321.html (B (Bmack_jpn (B (BOn Thu, 1 Apr 2004 01:08:44 +0900 $B4dED(B $B?-2p(B <[EMAIL PROTECTED]> wrote: (B (B> hi. (B> (B> I got these compile errors while install asterisk. (B> readli

[Asterisk-Users] Virbiage Phones - Vapourware??

2004-03-31 Thread Aaron Martin
Has anyone heard any more info about the Virbiage FT201 VoIP phones?   About 3 months ago I was told they were 6 weeks away, about 3 weeks ago I was told they were 2 weeks away, and now I am told they are 2 months away again!  Are they EVER going to arrive?  Can anyone shed some light on thi

RE: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Gene Kochanowsky
This is the version with the problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, March 31, 2004 11:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Extension ringing but no ringing sound. On Wed, 2004-03-31 at 2

RE: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Eric Wieling
On Wed, 2004-03-31 at 20:57, Gene Kochanowsky wrote: > Thanks Eric. Is this the CVS branch you are referring to? - > > # cvs checkout -r v1-0_stable asterisk Yes. -- Useful Asterisk Docs: http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip

[Asterisk-Users] make asterisk ignore password in sip register

2004-03-31 Thread Greg Hill
I am looking for a way to set my * so that it'll accept REGISTERs with a particular username, regardless of the password used. Snooping the network traffic suggests that the password is sent in digest or some other non-plaintext format, so I don't suppose there is an easy way for me to find out wha

RE: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Gene Kochanowsky
The v1-0_stable version is the one with the problem. I have checked out the latest and it does ring. Gene -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gene Kochanowsky Sent: Wednesday, March 31, 2004 9:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk

RE: [Asterisk-Users] Re: Can't compile asterisk.

2004-03-31 Thread $B4dED(B $B?-2p(B
> This error probably indicates something significant. What (B> sort of system are you compiling on? It's quite unusual for (B> "pwd" to be missing on any unix-like system. So it must not (B> be in your search path. It's almost always /bin/pwd, and if (B> you don't have /bin and /usr/bin

RE: [Asterisk-Users] Newbie....

2004-03-31 Thread Hall, Eric M.
Could I do things like call other ext on the system? Check Voice mail? I would like to test this before I put money in cards I may not need. What Software Phone app is people using? Thanks for all the help so far. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On B

RE: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Gene Kochanowsky
Thanks Eric. Is this the CVS branch you are referring to? - # cvs checkout -r v1-0_stable asterisk Gene -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, March 31, 2004 8:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asteris

Re: [Asterisk-Users] C7960 "busy" notification

2004-03-31 Thread Bill Hamel
Perhaps the 7960 has "Call Waiting" set to "YES" - this being the case, you're really not hitting a "busy" but the 15 sec "timeout" instead. Try setting "CallWaiting=No" in the 7960 you should then get a 'busy'. Oh, another thing. If you have multiple line appearences configured as the same SIP p

Re: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Eric Wieling
On Wed, 2004-03-31 at 19:05, Gene Kochanowsky wrote: > Greetings, > > This is probably some configuration issue, but for some reason my system > has stopped playing a ringing sound when an extension is dialed. The > phone rings but there is no ring sound in the ear piece. You updated your Asteris

RE: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Gene Kochanowsky
Hi Mark, I don't think it is codec mangling. My entire system is TDM. On incoming calls, after asterisk picks up there is a ring but that is the only time I hear it. Gene -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday,

Re: [Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread kc2eni
Hi Gene, I had this problem. I found that it would work between devices that shared the same codec settings but not otherwise. Try adding |T to the end of your extensions dialing string. This will force * to handle the call and so do any codec mangling. This has a hit on your server but at least

Re: [Asterisk-Users] Voicemail prompts garbled

2004-03-31 Thread Andres
This has been discussed before. Insert a "wait" in your dialplan before the voicemail starts pumping out the rtp stream. It seems the Cisco phones are slow to set up the audio stream. Michael Welter wrote: I'm having problems with the voicemail prompts. The beginning of each prompt is garbl

[Asterisk-Users] Extension ringing but no ringing sound.

2004-03-31 Thread Gene Kochanowsky
Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky ___ Asterisk-Users mailin

[Asterisk-Users] safe_asterisk with non-root user

2004-03-31 Thread Ryan Courtnage
Hello, I've found a couple of previous posts on this subject, but with no posted resolution... I'm attempting to run * as a non-root user (asterisk), following the guidelines on the wiki: http://voip-info.org/tiki-index.php?page=Asterisk%20non-root I can run * as my new user with "/usr/sbin/

Re: [Asterisk-Users] SoftFAX/spandsp - txfax

2004-03-31 Thread Steve Underwood
Hi Alex, Alex Zarubin wrote: Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension w

Re: [Asterisk-Users] ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file

2004-03-31 Thread ast
Please send me a copy On Thu, 1 Apr 2004, Leo Ann Boon wrote: > Figured this might be interesting to some on the list. I've written a > small tool to combine the 2 res_monitor created wav files into a single > compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can > select the part

[Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?

2004-03-31 Thread Zot O'Connor
How do I do this 1) ZAP-> * -> IAX(1) --> IAX(2) -> DG104S --> Handset 2) No Answer on Handset 3) Back to IAX(1) 4) IAX(1) tries a cell phone 5) Still no Answer 6) Local * Voicemail. I have 1 working, and I had 4 working when there was only one box, i.e. when the handset did not answe

[Asterisk-Users] ANNOUCEMENT: Tool to combine call recording into a single 'stereo' file

2004-03-31 Thread Leo Ann Boon
Figured this might be interesting to some on the list. I've written a small tool to combine the 2 res_monitor created wav files into a single compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can select the party to listen to by fiddling with the audio balance. Email me off-list if

RE: [Asterisk-Users] sip-msmessenger

2004-03-31 Thread Girish Gopinath
Hello, From: Shawn <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip-msmessenger Date: 31 Mar 2004 17:03:08 -0500 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.100:9082 From: ;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: ;t

[Asterisk-Users] Voicemail prompts garbled

2004-03-31 Thread Michael Welter
I'm having problems with the voicemail prompts. The beginning of each prompt is garbled. Prompts "edian mail" and "assword" can be recognized, but when it starts on the short phrases it is completely garbled. It is almost like the subsequent phrase starts before the current phrase finishes.

Re: [Asterisk-Users] G726 not working ?

2004-03-31 Thread Bill Hamel
Hi, Thanks for the reply! I am still having troubles I did try: disallow=all allow=g726 And still get: Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible codecs! -b Quoting Michael Manousos <[EMAIL PROTECTED]>: > Hi, > > Bill Hamel wrote: > > Hi, > > > > I a

RE: [Asterisk-Users] setting up 7940

2004-03-31 Thread Simon Brown
The password has to match what you have in sip.conf Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Sent: Thursday, 1 April 2004 4:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] setting up 7940 Simon Brown wrote: >Make sure that you

[Asterisk-Users] Re: Newbie....

2004-03-31 Thread Hans-Henrik Andresen
> If you want > MusicOnHold and Conferencing however, you will need one card for the timing. Why - I had used only ztdummy that works for MOH and conf. uncomment ztdummy in the makefile for zaptel and compile. /Hans-Henrik Andresen ___ Asterisk-User

[Asterisk-Users] sip-msmessenger

2004-03-31 Thread Shawn
Can anyone please help, I can't tell why it will not connect. I do not know how to read this debug file to were it is wrong. Thanks Sip read: REGISTER sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:9082 From: ;tag=97442d5b-75b7-4e23-9021-b8605797eb56 To: Call-ID: [EMAIL PROTECTED] CSeq

Re: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Nate Carlson
On Wed, 31 Mar 2004, Angus Berry wrote: > Does anyone hook in to Vonage directly without their Cisco box.. I see > they have a soft phone out now. I couldn't get the Vonage softphone working for outgoing calls (incoming worked ok); need the SIP proxy support mentioned here: http://bugs.digium.co

Re: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Angus Berry
I agree... this gem of a link could be more prominent. I didn't find anything on the WiKi where folks comment about their 3rd party VoIP providers...did I miss something else? Does anyone hook in to Vonage directly without their Cisco box.. I see they have a soft phone out now. On Wed, 2004-03-3

Re: [Asterisk-Users] setting up 7940

2004-03-31 Thread kwijibo
Are you trying to change the password on the phone or the password it sends to Asterisk to authenticate with? To change the password to the phone you need to put: phone_password: "somepassword" in either the SIP.cnf or the SIPDefault.cnf, depending how global you wish to make the password. Remem

[Asterisk-Users] VON show report

2004-03-31 Thread James H. Thompson
I wrote up report on products that caught my interest at the VON show going on this week in Santa Clara.       http://www.voip-info.org/wiki-VON+Spring+2004+Report     Jim   James H. Thompson[EMAIL PROTECTED]

Re: [Asterisk-Users] Answering Machine Detection

2004-03-31 Thread Peter Brown
"Francois Lambert" <[EMAIL PROTECTED]> At 13:15 31/03/04 +0530, you wrote: Hi, How do I detect an Answering Machine in Asterisk. I saw a post by Francois Lambert on 19 Jan. but am unable to get his email id. http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html Can somebody please help? Tha

Re: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Walt Reed
On Wed, Mar 31, 2004 at 03:06:21PM -0500, Angus Berry said: > thanks... this makes for great reading. > > Is there a link from Asteriskpbx.org that I missed? It IS there... Go to support, and it's the first under "user contributed links". Considering how valuable this Wiki is, it would be nice to

Re: [Asterisk-Users] SMDI support in Asterisk ?

2004-03-31 Thread Dave Packham
We are looking into using the Linux SMDI code to write a * app module. have not gotten far.. but we could help... with the Linux SMDI stuff already written it shouldn't be too hard. http://rpmfind.net/linux/RPM/contrib/libc6/i386/smdi-0.0.3-1.i386.html like that Dave P >>> [EMAIL PROTECTED]

Re: [Asterisk-Users] C7960 "busy" notification

2004-03-31 Thread Eric Wieling
You would need to turn off call waiting on the phone. I think there's a menu option or config option for the phone for that. On Wed, 2004-03-31 at 11:57, Rich Adamson wrote: > Using the following defnitions with a C7960: > > exten => 3001,1,Dial(SIP/3001,15,r) > exten => 3001,2,Voicemail2(u3001)

[Asterisk-Users] Re: X100P Echo was: USB Headsets (Plantronics DSP-400)

2004-03-31 Thread Doug Meredith
"Jason A. Pattie" <[EMAIL PROTECTED]> wrote: >Is there any possibility to remove the "turnaround" leg or whatever its >called at the X100P? I'm just thinking of a scenario where none of the >outgoing signal is ever introduced to the incoming circuit. That way, >the echo problem simply disappears

Re: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Angus Berry
thanks... this makes for great reading. Is there a link from Asteriskpbx.org that I missed? On Wed, 2004-03-31 at 14:05, Walt Reed wrote: > On Wed, Mar 31, 2004 at 01:06:03PM -0500, Angus Berry said: > > It would be useful to have a WiKi on asteriskpbx.com where user could > > post their experien

RE: [Asterisk-Users] Newbie....

2004-03-31 Thread Hall, Eric M.
Now for the next question. I have an old AT&T Merlin Mail system with a Brooktrout comcode series 4 cards in it. Could I use them? Thanks again for your help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Gudino Sent: Wednesday, March 31, 2004

Re: [Asterisk-Users] Newbie....

2004-03-31 Thread Nicolas Gudino
Hi, On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote: > I have a question for the group. > To get this running do I need any Digium Cards? I understand I will > need them to connect to the public phone system. I'm looking at just > using IP Phones or IP Softphones just to test this app. You can

RE: [Asterisk-Users] Newbie....

2004-03-31 Thread Brian Mulligan
Correction to my previous post. You do not need a Digium card for MusicOnHold just the conferencing...eeuuurgghh Brian > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Hall, Eric M. > Sent: 31 March 2004 20:01 > To: [EMAIL PROTECTED] > Subject: [Asterisk

Re: [Asterisk-Users] Newbie....

2004-03-31 Thread Iain Stevenson
--On Wednesday, March 31, 2004 2:00 pm -0500 "Hall, Eric M." <[EMAIL PROTECTED]> wrote: I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones jus

RE: [Asterisk-Users] Newbie....

2004-03-31 Thread Brian Mulligan
You can get by without Digium cards if all you want is IPPhones. If you want MusicOnHold and Conferencing however, you will need one card for the timing. I am told that you can emulate this hardware feature but with a basic FXO card costing only 60 quid, why bother. You can find an excellent descri

Re: [Asterisk-Users] Voicemail Options

2004-03-31 Thread Scott Laird
On Mar 31, 2004, at 10:12 AM, Steven Critchfield wrote: On Wed, 2004-03-31 at 11:51, Ryan Thrash wrote: How do I set configure my voicemail notification so that when I'm left a voicemail message it: 1) sends an e-mail to my inbox with the voicemail message attached 2) sends a message to my cellphon

RE: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Senad Jordanovic
> > A quick hunt around the net shows this Ericsson unit on E-bay as > H.323 only. The price is good, but I'd rather have SIPactually > I'd rather have IAX2! > > Michael Yeah.. It is pain to set it up... But it does work very well... And.. IAX2 box... :) would be very very nice.. Mark? A

Re: [Asterisk-Users] Basic Answering Machine Function?

2004-03-31 Thread Scott Weis
Here is how I do the same thing: exten => 1234,1,Dial(Zap/2,30) exten => 1234,2,Answer exten => 1234,3,DigitTimeout,5 exten => 1234,4,ResponseTimeout,3 exten => 1234,5,SetMusicOnHold(random) exten => 1234,6,BackGround(2) exten => 1234,7,BackGround(vm-nobodyavail) exten => 1234,8,Voicemail(21) Thi

Re: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Walt Reed
On Wed, Mar 31, 2004 at 01:06:03PM -0500, Angus Berry said: > It would be useful to have a WiKi on asteriskpbx.com where user could > post their experiences of 'production' multi-line installs. > > I'm 'all ears' to how far folks have scaled their installs especially > with T1/E1 size systems. Yo

RE: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Lee Cremeans
We're running a T1-based system here (8 incoming lines, 7 of which are in a rollover group through our telco, 1 of which is our fax line) with a set of 10 Cisco 7940s and a 7960 with the SIP firmware, and so far Asterisk and the T100P have worked great, modulo a few problems with slightly dodgy har

[Asterisk-Users] Newbie....

2004-03-31 Thread Hall, Eric M.
I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give. _

[Asterisk-Users] SER Asterisk problem

2004-03-31 Thread Welesley Sibelson Dias
Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&

[Asterisk-Users] noise in a single direction call

2004-03-31 Thread Marko Rakar
We have asterisk server which have hfcs isdn adapter acting as FXO and mediatrix units as FXS on another site connected with 2mbit link. The trouble is that when I connect from asterisk server to mediatrix unit; whatever comes to asterisk is heard OK (no noise, minimum jitter; absolutely acceptab

[Asterisk-Users] Basic Answering Machine Function?

2004-03-31 Thread Jeff Rush
I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendan

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Michael Graves
On Wed, 31 Mar 2004 12:54:02 -0500, Angus Berry wrote: >A quick search on eBay turned up this 4 port FXO external box for >US$299: > >http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=51279 > >...anyone know if it's compatible with Asterisk? > >Tip: Search eBay for "FXO" and c

RE: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Angus Berry
So is this the current best price solution for 4 port FXO... and no PCI slots used? On Wed, 2004-03-31 at 13:26, Senad Jordanovic wrote: > Angus Berry wrote: > > A quick search on eBay turned up this 4 port FXO external box for > > US$299: > > > > > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&i

[Asterisk-Users] (no subject)

2004-03-31 Thread jay
unsubscribe ___ 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] 3-4 port FXO card recommendations

2004-03-31 Thread Senad Jordanovic
Angus Berry wrote: > A quick search on eBay turned up this 4 port FXO external box for > US$299: > > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=5 1279 > > ...anyone know if it's compatible with Asterisk? Yes.. I can confirm I had it setup and it is working great. ___

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Gregory Junker
In my case, because the server I have decided to use for Asterisk has most of the slots filled with other cards already. I'm sure my situation is fairly typical. Greg On Wed, 2004-03-31 at 11:22 -0600, John Baker wrote: > Why not just get a motherboard with 6 pci slots? I don't know your > set

RE: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Angus Berry
It would be useful to have a WiKi on asteriskpbx.com where user could post their experiences of 'production' multi-line installs. I'm 'all ears' to how far folks have scaled their installs especially with T1/E1 size systems. On Wed, 2004-03-31 at 12:09, jc wrote: > Well, I hope it works because w

Re: [Asterisk-Users] Voicemail Options

2004-03-31 Thread Steven Critchfield
On Wed, 2004-03-31 at 11:51, Ryan Thrash wrote: > How do I set configure my voicemail notification so that when I'm left > a voicemail message it: > > 1) sends an e-mail to my inbox with the voicemail message attached > 2) sends a message to my cellphone without the message attached Use procmail

[Asterisk-Users] WTS (200) Cisco ATA-186-I1

2004-03-31 Thread Sales Department
Condition: New Open Box Warranty: 90 Days Cost - $130/ea Minimum Order 5pcs Contact [EMAIL PROTECTED] for details Cory Andrews ++ b2 technologies 454 Sonwil Drive Buffalo, NY 14225 ++ email - [EMAIL PROTECTED] voice - 716.630.1555 X22 fax - 716.630.1548 web - www.ValueRe

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Angus Berry
A quick search on eBay turned up this 4 port FXO external box for US$299: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=51279 ...anyone know if it's compatible with Asterisk? Tip: Search eBay for "FXO" and check the box saying search titles and descriptions. On Wed, 200

Re: [Asterisk-Users] setting up 7940

2004-03-31 Thread Roger
Simon Brown wrote: Make sure that you have the following (or equivalent in the SIP.conf # Line 1 appearance line1_name: 202 # Line 1 Registration Authentication line1_authname: "202" # Line 1 Registration Password line1_password: "202" and this in the SIPDefault.conf # Proxy Registration (0-dis

[Asterisk-Users] C7960 "busy" notification

2004-03-31 Thread Rich Adamson
Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second ti

[Asterisk-Users] Voicemail Options

2004-03-31 Thread Ryan Thrash
How do I set configure my voicemail notification so that when I'm left a voicemail message it: 1) sends an e-mail to my inbox with the voicemail message attached 2) sends a message to my cellphone without the message attached I get notifications when I've got attachments turned off, but my cell

RE: [Asterisk-Users] Need help understanding SIP phones

2004-03-31 Thread Lee Cremeans
>How would/do vendor specific phones like cisco 7900's and 3Com VCX 3100 >series phones work SIP to SIP if they download the protocol from the >Vendor call centers? At least in the 79xx's case, the phone actually downloads firmware updates from a TFTP server; by changing the name of the image to u

Re: [Asterisk-Users] Manager Interface "Action: Originate" changed

2004-03-31 Thread Tony Wasson
Tony Wasson wrote: I have recently noticed that the "Action: Originate" options in asterisk 1.0 CVS has changed sometime between 2/23 and 3/18. To post a follow up for posterity, 2 tips were suggested when using the Manager Interface: 1) Make sure to supply Context AND Priority when using

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread John Baker
Why not just get a motherboard with 6 pci slots? I don't know your setup, but you could free up a slot (or two) by buying a motherboard with a built-in lan (and/or sound). John Ariel Batista wrote: Mark Buckaway wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro

RE: [Asterisk-Users] ANNOUNCEMENT : MeetMe Web User Interface

2004-03-31 Thread Dan Austin
Very cool. With the changes to move the MeetMe data into a DB, would it be possible to add scheduling, password management and caller limits to the app? As someone else pointed out this is an almost perfect fit for our conferencing needs. Dan -Original Message- From: Areski [mailto:[EM

Re: [Asterisk-Users] X100P Echo was: USB Headsets (Plantronics DSP-400)

2004-03-31 Thread Jason A. Pattie
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brent Franks wrote: | Agreed. It seems that one out of about every 40 calls has echo problems | for the duration of the cell. I also notice it a lot more when the call | is incoming vs. outgoing. But it will still occur on outgoing calls | occasional

RE: [Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread jc
Well, I hope it works because we are installing one in a new office tomorrow! Ill let you know how it goes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Mangin Sent: Wednesday, March 31, 2004 5:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-User

[Asterisk-Users] Config file references (was g726 not working)

2004-03-31 Thread brook davis
So in my sip.conf I put many variations of what I thought should go in there, finally including (to no avail): I have had to play this guessing game as well with other codecs/config file settings in general. And I think this email touches on something that has been troubling me for sometime.

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Jason Becker
Scott Laird wrote: On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote: In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that suppor

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3276 - 7 msgs

2004-03-31 Thread Darnell Gadberry
Sam, I too am experiencing a similar problem with the VoicePulse LA exchange. I have two VoicePulse DID numbers 212 XXX and 213 XXX. Both numbers terminate in the same Asterisk context on my server. DTMF is being passed-through properly on the 212 number but not the 213. I have sent VoicePulse

Re: [Asterisk-Users] Hangup not detected on X100P

2004-03-31 Thread Steven Critchfield
On Wed, 2004-03-31 at 10:14, Matt Bridges wrote: > I've configured my [*] to dial the pstn which is working like a charm. > I've also configured an extension to ring when the PSTN line is ringing > which is also working brilliantly, but, sometimes it doesn't detect that the > call has been hungup.

Re: [Asterisk-Users] Play file from an offset

2004-03-31 Thread Steven Critchfield
On Wed, 2004-03-31 at 10:06, Yves Chouinard wrote: > I am projecting to migrate applications written with the Dialogic API > to Asterisk. There are a few things that I do with Dialogic that I am > still not sure are possible with Asterisk : > > - play a file from an offset (so a user can press a

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Gregory Junker
Yay!! Precisely what I need as well (4-port)...do these rumours include any word on timeframe? Greg On Wed, 2004-03-31 at 08:31 -0800, Scott Laird wrote: > On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote: > > In setting up Asterisk, I'm looking to dump my current phone system > > (Nortel Ventur

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Michael Graves
On Wed, 31 Mar 2004 08:31:38 -0800, Scott Laird wrote: > >On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote: >> In setting up Asterisk, I'm looking to dump my current phone system >> (Nortel Venture). I presently have three POTS lines. >> >> I would use a VOIP provider, but now are presently avail

Re: [Asterisk-Users] Hangup not detected on X100P

2004-03-31 Thread WipeOut
Matt Bridges wrote: I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Ariel Batista
Mark Buckaway wrote: > *This message was transferred with a trial version of CommuniGate(tm) > Pro* > In setting up Asterisk, I'm looking to dump my current phone system > (Nortel Venture). I presently have three POTS lines. > > I would use a VOIP provider, but now are presently available in the >

[Asterisk-Users] SMDI support in Asterisk ?

2004-03-31 Thread tony banks
Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

[Asterisk-Users] Re: Can't compile asterisk.

2004-03-31 Thread John Chambers
$B4dED(B $B?-2p(B wrote: (B (B> I got these compile errors while install asterisk. (B> readline and openssl are compiled using gnu source, and kernel version is 2.4.17. (B> (B> Compile errors message is follows. (B> Someone cleared this problem? (B> Please, help! (B> (B> Regards. (B

Re: [Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Scott Laird
On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote: In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware

Re: [Asterisk-Users] Can't compile asterisk.

2004-03-31 Thread Dpto. Tecnico.
The problems are that you don't have instaled several shell commands like: (Bpwd (BThe logs told you in the 7th line. (B (BHave a nice day. (B (B- Original Message - (BFrom: "$B4dED(B $B?-2p(B" <[EMAIL PROTECTED]> (BTo: <[EMAIL PROTECTED]> (BSent: Wednesday, March 31, 2004

Re: [Asterisk-Users] setting up 7940

2004-03-31 Thread Roger
Michael Welter wrote: You phone isn't getting to the tftp server or the tftp server doesn't have the required files. Make sure your dhcp server is specifying the correct IP of the tftp server. You can also check the phone's parameters to verify the tftp address. If that is ok then you need a

RE: [Asterisk-Users] Play file from an offset

2004-03-31 Thread Pedro Bessa Goncalves
Hi. I am also doing same thing as you. Currently my company has a PBX platform which uses Dialogic hardware to control channels. Now they want their software to work over Asterisk. I still haven’t started implementing modifications, as I am still studying Asterisk. I think it would be inte

[Asterisk-Users] Re: Reliable Provider

2004-03-31 Thread Kurt Pasewaldt
Net2phone Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asteri

[Asterisk-Users] Hangup not detected on X100P

2004-03-31 Thread Matt Bridges
I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf file

[Asterisk-Users] Re: ATA registration requests

2004-03-31 Thread Kurt Pasewaldt
I set the SISIPRegIntervalo 3600. Wouldn't that mean to send a registration packet every hour instead of every miminuter so. --OR-- Is this the typical reresponseack to the * server when the ATATAeceive a SIP Notify. Kurt __

[Asterisk-Users] Asterisk as office PBX

2004-03-31 Thread Thomas Mangin
Hello, I would like to make sure I am not going to shoot myself in the head, so I would appreciate if you could tell me if what I am willing to do is realistic (and will work) or not. I am willing to replace our office "London" phone system when a new E1 (G703) is installed later this month and I

[Asterisk-Users] Play file from an offset

2004-03-31 Thread Yves Chouinard
I am projecting to migrate applications written with the Dialogic API to Asterisk. There are a few things that I do with Dialogic that I am still not sure are possible with Asterisk :   - play a file from an offset (so a user can press a key to rewind 3 sec., pause, etc.) - dynamic volume c

[Asterisk-Users] 3-4 port FXO card recommendations

2004-03-31 Thread Mark Buckaway
*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that su

[Asterisk-Users] Need help understanding SIP phones

2004-03-31 Thread Jeff Donovan
Greetings i have been using asterisk with good success with soft phones and I need to purchase some Sip capable phones to test. How would/do vendor specific phones like cisco 7900's and 3Com VCX 3100 series phones work SIP to SIP if they download the protocol from the Vendor call centers? I w

Re: [Asterisk-Users] RE: RxFax/spandsp: not disconnecting

2004-03-31 Thread Steve Underwood
Reynaldo Simbulan wrote: Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. spandsp-0.0.1k.tar.gz and updated

[Asterisk-Users] Can't compile asterisk.

2004-03-31 Thread $B4dED(B $B?-2p(B
hi. (B (BI got these compile errors while install asterisk. (Breadline and openssl are compiled using gnu source, and kernel version is 2.4.17. (B (BCompile errors message is follows. (BSomeone cleared this problem? (BPlease, help! (B (BRegards. (B (B-

[Asterisk-Users] Asterish<->Cisco Call manager

2004-03-31 Thread Mark Buckaway
*This message was transferred with a trial version of CommuniGate(tm) Pro* Does anyone have experience with connect Asterish to Cisco Call Manager 3.x? What I would like to try to do is to connect my home asterish into the office Cisco call manager most likely over a VPN from home to the office T

Re: [Asterisk-Users] ANNOUNCEMENT : MeetMe Web User Interface

2004-03-31 Thread WipeOut
Looks like a cool system.. looking forward to seeing it develop.. Later.. Areski wrote: Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual pres

Re: [Asterisk-Users] H323 in Asterisk

2004-03-31 Thread Michael Manousos
Terence Parker wrote: I have posted before but didn't get any replies so i'll ask again in a more simple way : Does H323 work on asterisk out of the box? I notice there is already a channels/chan_h323.c file, but creating an h323.conf file I can't seem to get H323 working. Do I have to compi

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