Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

At 11:20 11/01/04 +0800, you wrote: Anton Tinchev wrote: Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks Unless one ha

Re: [Asterisk-Users] Asterisk Development Updates

On Sat, 10 Jan 2004, Jared Smith wrote: > On Sat, 2004-01-10 at 20:25, Greg Boehnlein wrote: > > > Awesome! I'm game to create Asterisk RPMS when the stable branch comes > > out! > > Great... I was going to do the same... maybe we should join forces and > make better RPMS! (I've already got a

Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */

Anyone who does not like the fact that all code must be "disclaimed", sent through Digum to CVS and that GPL'd code can't go in can fix that problem. All you need to do is copy the current CVS and use that to start your own project. You can call it "Asterisk Prime" or "Star" and make up your own

RE : [Asterisk-Users] 2nd call leg status?

Hi Folks, Wonder whether this question found an answer ? I too have a similar question that I can't find an answer so far. Let me first share my dial plan; exten => _011.,1,Authenticate(/etc/asterisk/auth.txt |a) exten => _011.,2,Playback(Pls-wait-while-I-connect) exten => _011.,3,Absolutetimeo

[Asterisk-Users] a constructive proposal: tie the marshals to a cvs server

perhaps i have been too ascerbic on the list as of late. for that i offer my apologies. create a second cvs server, bug marshalls have commit access. they commit patches (tagged with bug id) to this server. users wanting bleeding-edge stuff, use that server and don't have to keep downloading and

Re: [Asterisk-Users] default music source for SIP channel

Lance Arbuckle wrote: Thanks Kevin, but boy, do I feel dumb. Maybe someone could update the MusicOnHold wiki page and add SetMusicOnHold to the "Also See" section. Added to wiki and submitted a patch with new "show application musiconhold" help text to reflect this. /O __

Re: [Asterisk-Users] default music source for SIP channel

Lance Arbuckle wrote: So, why does zapata.conf accept musiconhold=class yet sip.conf ignores a similar statement ? Can anyone give me an example of how to control the MOH class for a SIP channel ? Because no one added that feature to chan_sip.c For now, use setmusiconhold, then you're welcome t

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

Chandra said: > i also had the same problem temporarily i solved my problem with both > outside NAT. u can also do it if both inside NAT. * outside NAT and > Budgetone behind NAT simply doesn't seem to work. if u ever solve this > problem please let me know too. > > thanks > > cm > I am able t

Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

Hi All, i just applied this patch. i need to test whether its working. Can someone connect to my server and leave me a vm at extension 2000. Server : ojoobala.com Phone Extension : 2005 pwd : mytest auth: md5. pl leave a vm on extension 2000. thanks a lot, -B - Original Messa

Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

Peter Brown wrote: At 11:20 11/01/04 +0800, you wrote: Anton Tinchev wrote: Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB.

Re: [Asterisk-Users] Asterisk behind NAT << How to do it.

i like the idea of not requiring to open 1 ports in the firewall. Do i need to change rtf.conf to from 1 - 2 to 16384 and 16394. thanks, -B - Original Message - From: "Craig Waddington" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, December 27, 2003 3:43 AM Sub

[Asterisk-Users] High Level of CVS activity

I don't think I've ever seen so many commits to the CVS, I'm not complaining. To save lots of checkout/test cycles can a message be posted when the current surge is over? -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene 04 90 23 30 81 IA

RE: [Asterisk-Users] Asterisk behind NAT << How to do it.

Balaji. I just left rtf.conf at default. Though I guess it wouldn't hurt to change it to test. Does it currently work for you with the settings I provided? Craig. www.ntfs.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: 11 Janua

Re: [Asterisk-Users] default music source for SIP channel

"Olle E. Johansson" wrote: > > Lance Arbuckle wrote: > > > > > > So, why does zapata.conf accept musiconhold=class yet sip.conf ignores a > > similar statement ? Can anyone give me an example of how to control the > > MOH class for a SIP channel ? > Because no one added that feature to chan_sip

Re: [Asterisk-Users] far end disconnect supervision

> > Personal opinion is that throwing a channel bank at * (other then for > > better echo cancellation on trunks) is like taking a shower with your > > socks on; don't see any practical use. But, I'm sure some do. > > This statement assumes a lot. > > 1. That you don't want to use analog phones

[Asterisk-Users] More Success on the Cisco 7920 and SCCP !!!!!

Hi All, have some decent success on the 7920 "activation" in Asterisk. Latest status: chan_skinny does NOT work with 7920 chan_sccp does WORK with 7920 (!!) however: to remove coredumping the chan_sccp just comment out the MWI (messagewaitingindicator), then it compiles fine. Then change sccp_

[Asterisk-Users] Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!

Hi Jan, the 7920 is on my todo list for quite a few days now, and I've had experience similar to yours... On Sun, 11 Jan 2004, Jan Czmok wrote: > Latest status: > chan_skinny does NOT work with 7920 > chan_sccp does WORK with 7920 (!!) Yup. One should add that you'd better use the 0.2 release o

Re: [Asterisk-Users] High Level of CVS activity

On Sun, 11 Jan 2004, Dave Cotton wrote: > I don't think I've ever seen so many commits to the CVS, I'm not > complaining. To save lots of checkout/test cycles can a message be > posted when the current surge is over? Most likely that is the frenzy before the release of 0.7.0 on Monday, the 12th!

Re: [Asterisk-Users] Development Process comment and Email list suggestion

On Fri, 9 Jan 2004 [EMAIL PROTECTED] wrote: [...] > Regarding the email list: A number of people have suggested creating more > email lists. I think this is not a good idea because there will be even > more cross posting than there is now between -dev and -users. That's a very valid point. [...]

Re: [Asterisk-Users] far end disconnect supervision

Hi! > I've not implemented any form of channel bank with *, so can't offer much > help on specific vendor/models. Since there are a fair number of folks > using them on the list, try posting a new thread with channel bank in > the subject, and summarize the responses in the wiki. Yes, please do s

Re: [Asterisk-Users] default music source for SIP channel

Hi! > Thanks Kevin, but boy, do I feel dumb. Maybe someone could update the > MusicOnHold wiki page and add SetMusicOnHold to the "Also See" > section. Just login to the Wiki and do that yourself. That's the idea of a Wiki! Greez, Philipp ___ Aster

[Asterisk-Users] New Version of SJPhone

I just installed the new version of SJPhone and it appears that it cannot work with * anymore?

RE : [Asterisk-Users] 2nd call leg status?

Hi Folks, Wonder whether this question found an answer ? I too have a similar question that I can't find an answer so far. Let me first share my dial plan; exten => _011.,1,Authenticate(/etc/asterisk/auth.txt |a) exten => _011.,2,Playback(Pls-wait-while-I-connect) exten => _011.,3,Absolutetimeo

[Asterisk-Users] Cisco 79xx Ringtones

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! An

[Asterisk-Users] Strange problem with call hangup on Budgetone 102 Phones

Hi,   I've got Asterisk configured and working (sort of) with an Eicon Diva Server 2M ISDN card (connected to S0 bus of another PBX). This * box is on a 'live', non-nat IP address. I also have a couple of budgetone phones, one behind NAT and one not. When I place an outgoing call, I get the f

[Asterisk-Users] macro error "exited non-zero"

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [EM

Re: [Asterisk-Users] High Level of CVS activity

On Sunday 11 January 2004 04:36, Dave Cotton wrote: > I don't think I've ever seen so many commits to the CVS, I'm not > complaining. To save lots of checkout/test cycles can a message be > posted when the current surge is over? Yes, there were a few of us going through the bugtracker last night a

[Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

Dell - PowerEdge 400SC Server Under $300 with MIR Intel® P®4 Processor at 2.8GHz, 512KB Cache, 800MHz FSB For those of you who aren`t familiar with the 400SC, this server is an Intel i875P chipset based server with an 8x AGP slot. It is compatible with 533Mhz and 800Mhz processors (hyperthread

Re: [Asterisk-Users] High Level of CVS activity

On Sun, 2004-01-11 at 18:00, Tilghman Lesher wrote: > Yes, there were a few of us going through the bugtracker last night > and trying various patches for the 0.7.0 release. The ironic thing is > that approximately the same time that you posted is the same time > that a whole lot of the trackers q

Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

On Sun, 2004-01-11 at 18:06, Richard Grinnell wrote: > Dell - PowerEdge 400SC Server Under $300 with MIR > Intel P4 Processor at 2.8GHz, 512KB Cache, 800MHz > FSB Yet again the trans-Atlantic con 399¤ here = $500 when it should be 235¤ Just paid 5000¤ for a 2600SC but it is nice, if only it did

RE: [Asterisk-Users] Cisco 79xx Ringtones

Customizing the Cisco SIP IP Phone Ring Types The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By default, your ring type options will be those two choices. However, using the RINGLIST.DAT file, you can customize the ring types that are available to the Cisco SIP IP phone users

RE: [Asterisk-Users] Cisco 79xx Ringtones

http://www.bkw.org/~brian/cisco/7960-ringtones/rings/ bkw On Sun, 11 Jan 2004, Craig Waddington wrote: > Customizing the Cisco SIP IP Phone Ring Types > > The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By > default, your ring type options will be those two choices. However,

Re: [Asterisk-Users] Need 3.3v 64 PCI Booard with 800FSB (for ordinary Pentium4, not Xeon)

Got one, and it is rock solid and totally quiet. Had an Athlon XP1800+ before with a screaming 7k RPM fan. Never even knew the clock on the wall actually ticked until I shutdown that noisemaker and fired up this 400sc. It is a bit picky about memory to take advantage of the bus speed, stick with th

Re: [Asterisk-Users] macro error "exited non-zero"

On Sunday 11 January 2004 11:47, Jimmy Riley wrote: > On this macro I keep getting exited non-zero on s,3, but s,3 is doing > what it is suppose to do but the macro stops. Is there a way to make a > macro ignore errors and continue to s,4? I have the lattes ver of sox > 12.17.4. Are you using Re

Re: [Asterisk-Users] Asteriks as SIP<>H323 Proxy?

On Sat, 10 Jan 2004, Arnd Vehling wrote: > is it possible to use Asteriks for translating SIP to H323 and vice versa? > I am looking to implement the following Setup > > SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC > > Basicly i want SIP fones to talk to H323 fones and an

RE: [Asterisk-Users] macro error "exited non-zero"

Jimmy Riley Network Administrator VeriCore 985-626-1701 X1103 -Original Message- From: Andres [mailto:[EMAIL PROTECTED] Sent: January 11, 2004 12:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] macro error "exited non-zero" On Sunday 11 January 2004 11:47, Jimmy Riley wrote: >

Re: [Asterisk-Users] far end disconnect supervision

Quoting Rich Adamson <[EMAIL PROTECTED]>: > > > Personal opinion is that throwing a channel bank at * (other then for > > > better echo cancellation on trunks) is like taking a shower with your > > > socks on; don't see any practical use. But, I'm sure some do. > > > > This statement assumes a l

RE : [Asterisk-Users] 2nd call leg status?

Thanks Freddi, Cool, it works, infact better yet ResetCDR(). ResetCDR() show just one record in cdr, ResetCDR(w) will crete two records with the one for first call leg, which I do not want :). Cheers SW Date: Sun, 11 Jan 2004 17:27:08 +0100 From: Freddi Hansen <[EMAIL PROTECTED]> To: [EMAIL PRO

[Asterisk-Users] SpeakFree

If you did not see slashdot today, check out this anoucment. http://www.fourmilab.ch/speakfree/ Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] macro error "exited non-zero"

On Sunday 11 January 2004 14:11, Jimmy Riley wrote: > Jimmy Riley > Network Administrator > VeriCore > 985-626-1701 X1103 > -Original Message- > From: Andres [mailto:[EMAIL PROTECTED] > Sent: January 11, 2004 12:31 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] macro error "exit

Re: [Asterisk-Users] Cisco 79xx Ringtones

Hi, On Sun, 11 Jan 2004, Adthrawn wrote: > I'm after two very specific ringtones for the 79xx's... > > A dog barking, and a horse either galloping or neighing. [...] > I do recall, you had to set the sample length to a divisible, something > like 800? And there was a maximum sample length too...

Re: [Asterisk-Users] SpeakFree

sad, yes but who needs speakfreely when you have asterisk and soft/hard phones. The author seems really unmotivated so let him find the path maybe he could join asterisk development team :) Miguel On Sun, 2004-01-11 at 19:33, Lists wrote: > If you did not see slashdot today, check out this anoucme

RE: [Asterisk-Users] SIP reload configuration problem /* New subject */

New users arent being added from the sip.conf file... let me play with it and get back to you in a few weeks when I know what I am doing! Newbie remember! Dont want to send you down the wrong track and then work out that its me doing something wrong. Thanks for your help Cheers Chris -Ori

[Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

We have a new contest starting today! The first three members to post 300 messages at http://www.asterisk.bz will win a _80Gig Hard Drive!_ Its quite simple. Messages must be asterisk related. http://www.asterisk.bz Alternative to the asterisk-users list

RE: [Asterisk-Users] SIP reload configuration problem /* New subject */

accually latest cvs has alot of minor and a few major fixes Worth the try.. its building up the 0.7.0 for monday. :) bkw On Mon, 12 Jan 2004, Christopher Raper wrote: > New users arent being added from the sip.conf file... > let me play with it and get back to you in a few weeks when I know

Re: [Asterisk-Users] SpeakFree

Since its public domain software we can use the encrypition part in IAX2 :) bkw On Sun, 11 Jan 2004, Miguel Cavazos wrote: > sad, yes but who needs speakfreely when you have asterisk and soft/hard > phones. The author seems really unmotivated so let him find the path > maybe he could join asteri

RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

Oh, I thought it was a contest for top posters! Darn! David, it's a cool and clean format, but what's the matter with using the Wiki that people already have devoted much blood sweat and tears to? It's not too bad - try it! Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Email

RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 4:31 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > We have a new contest starting today! > > The fir

[Asterisk-Users] Asterisk on FreeBSD 4.9

Hi all!I'm trying to set up Asterisk on FreeBSD 4.9 to route calls to H.323 GK.I have installed asterisk using the ports.It seems to be running OK, but when i try to dial through h323, it segfaults.I'm using X-Lite as SIP client, i have set up my h323.conf:[general]port = 1721bindaddr = 0.0.0.0tos

[Asterisk-Users] Asterisk on FreeBSD 4.9?

Hi all I'm trying to set up Asterisk on FreeBSD 4.9 to route calls to H.323 GK. I have installed asterisk using the ports. It seems to be running OK, but when i try to dial through h323, it segfaults I'm using X-Lite as SIP client, i have set up my h323.conf [general port = 1721 bindaddr = 0.0.0

[Asterisk-Users] "friendly" dial tone frequency combinations

Hello List, I just started messing with the settings on a SPA-2000, and it has a really nasty alternative dial tone that I want to make go away. I'm not too hip on how the two numbers interact, so my results haven't been good. (I'm in the US, so I'm bias'ed towards US tones.) Default(I'm ok with

Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

Hi > http://www.asterisk.bz Alternative to the asterisk-users list nothing against this forum, but this made me think. I noticed that some people loose their time in setting up doc sites... the idea is great, but since there're already grown sites (oej's wiki), why not stopping into doing some

RE: [Asterisk-Users] Newbie Question-Looking for Feedback

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Christopher Raper > Sent: Thursday, 8 January 2004 10:06 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Newbie Question-Looking for Feedback > > Greetings all. I am new to the Asterisk worl

[Asterisk-Users] analog or sip ? was far end disconnect supervision

Thanks to everyone that responded to my channel bank question. Ive decided that the Adit 600 would be a good choice. Then I got to thinking about SIP phones and wondered if their quality has progressed to the point that they can be deployed to customers who "just want their phones to work" and wo

Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

hi. > Then I got to thinking about SIP phones and wondered if their quality > has progressed to the point that they can be deployed to customers who > "just want their phones to work" and wouldn't tolerate any SIP hickups. so for that use Cisco. beside I like GS budgetones and wanna see them wor

Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

Hi! > As for pricing, I would think the SIP phones would need to be in the > $200 price range to be competative with analog or ADSI phones That would make it SNOM then, I believe. Or go look at MGCP phones. By the way, is anyone here using the SNOM 100 or 105? If yes, could you drop a short no

[Asterisk-Users] Re: [Asterisk-Dev] More Success on the Cisco 7920 and SCCP !!!!!

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > C

Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

I think wiki is one of the best resources around. I think fourms could be an addidional resource for the newbie, an alternative to a listserv. Scott Stingel wrote: Oh, I thought it was a contest for top posters! Darn! David, it's a cool and clean format, but what's the matter with using the W

Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

Forums are more desirable for the newbie user. I realize people are set the their ways. thats fine :) [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Burr Sent: Sunday, January 11, 2004 4:31 PM To: [EMAIL PROTECTED] Subjec

Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

The following errors occurred during your registration: * The username you entered as your referrer could not be located. cant create a username Miguel On Sun, 2004-01-11 at 22:53, Brancaleoni Matteo wrote: > Hi > > http://www.asterisk.bz Alternative to the asterisk-users list >

[Asterisk-Users] T1 Sync clarification

Hi List, After reading a bunch of the docs, list post archives, it still seems that a clear definition of how to clock the T100P card is muddy. zttool says that the link is "INTERNALLY CLOCKED", does this mean the T100P is providing clock, or does this mean the T100P is getting clock from the T

Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

sip phones have alot of nice features and they really work, you can try some phones under $200 yes, but about the analog phones, people like to have there cordless phones, or there micky mouse phone or garfield phone so consider that. You loss some features but your customers get the phones they w

[Asterisk-Users] NuFone Network H323 configuration?

I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/[EMAIL PROTECTED]) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets

Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

> Thanks to everyone that responded to my channel bank question. Ive > decided that the Adit 600 would be a good choice. > Then I got to thinking about SIP phones and wondered if their quality > has progressed to the point that they can be deployed to customers who > "just want their phones to wor

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote: > I'm using Asterisk on a open server (no firewall or NAT) and trying to > communicate with a Grandstream BudgeTone 102 SIP phone which is behind > NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from CVS > about a week ago.

Re: [Asterisk-Users] Chagres Technologies, Inc

On Saturday 10 January 2004 10:22 pm, Sean Cheesman wrote: > time to take this off-list. Pleeze! -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good!

Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?

On Sunday 11 January 2004 16:18, NetOne Administrator wrote: > I'm trying to set up Asterisk on FreeBSD 4.9 to route > calls to H.323 GK. > I have installed asterisk using the ports. > It seems to be running OK, but when i try to dial through > h323, it segfaults I want you to look at the headers

[Asterisk-Users] More words for Allison

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket.

RE: [Asterisk-Users] T1 Sync clarification

On Sunday, January 11, 2004 5:41 PM, John Brown (CV) [SMTP:[EMAIL PROTECTED] wrote: > Hi List, > > After reading a bunch of the docs, list post archives, it > still seems that a clear definition of how to clock the T100P > card is muddy. > > zttool says that the link is "INTERNALLY CLOCKED", > >

[Asterisk-Users] zttool and errors

It appears that zttool doesn't actually report T1 span errors. If I inject BPV's, crc errors, framing errors, etc into a T1 span, the counters on zttool don't change. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/

Re: [Asterisk-Users] T1 Sync clarification

THank you. Thats what I thought it should be. Off to call the telco and tell them they are mucked up. On Sun, Jan 11, 2004 at 06:54:11PM -0600, Don Pobanz wrote: > On Sunday, January 11, 2004 5:41 PM, John Brown (CV) > [SMTP:[EMAIL PROTECTED] wrote: > > Hi List, > > > > After reading a bunch

RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of David Burr > Sent: Sunday, January 11, 2004 6:32 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive > > > Forums are more desirable for the newbie user.

Re: [Asterisk-Users] More words for Allison

Add Celsius Fahrenheit bkw On Sun, 11 Jan 2004, John Todd wrote: > > Here's the latest batch of words to get shipped out to Allison Smith. > Please submit reasonably small changes to me by tomorrow 10:00 AM > Eastern time, and I'll add them. > > As usual, donations to what will be a ~$110 USD ex

Re: [Asterisk-Users] More words for Allison

How about Dollars Cents Euros On Sun, Jan 11, 2004 at 07:36:45PM -0500, John Todd wrote: > > Here's the latest batch of words to get shipped out to Allison Smith. > Please submit reasonably small changes to me by tomorrow 10:00 AM > Eastern time, and I'll add them. > > As usual, donations

[Asterisk-Users] Forward call with response required to accept

I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward.   Example:   PSTN1 Calls * dials PSTN2     if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a

Re: [Asterisk-Users] More words for Allison

Sex drigs rock & roll ???:-) On Sun, 11 Jan 2004 19:20:11 -0600 (CST), Brian West wrote: >Add Celsius Fahrenheit > >bkw > >On Sun, 11 Jan 2004, John Todd wrote: > >> >> Here's the latest batch of words to get shipped out to Allison Smith. >> Please submit reasonably small changes to me by

Re: [Asterisk-Users] question re voicemail

I think this is the syntax you are looking for   [sip]exten => 5104112978,1,Dial(SIP/5104112978,20,tr)exten => 5104112978,2,Voicemail,u5104112978 exten => 5104112978,102,Voicemail,b5104112978   - Original Message - From: Jess Magnaye To: [EMAIL PROTECTED] Sent:

RE: [Asterisk-Users] More words for Allison

> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of John Todd > Sent: Monday, 12 January 2004 11:37 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] More words for Allison > > knots per hour I'm a land-lubber, but I think knots is a speed unit (lik

[Asterisk-Users] WTS (200) AC Power Adapters for Cisco 7910 / 7940 / 7960 IP Phones

Have (200) Brand New power cubes (AC Power Adapter with AC Cord) - compatible with Cisco CP-7910, CP-7940, CP-7960 and equivalent "G" models.   $25/ea - Minimum Purchase (10) Units.   Email [EMAIL PROTECTED] if interested.   Regards   Cory Andrews***b2 Te

[Asterisk-Users] possible solution to PRI T100P dropped call issue

To recap: T100P card wouldn't sync with the telco using line side clocking ( span=1,1,0.) Had to use internal clocking (span=1,0,0...) zttool showed Tx/Rx Levels as 0/ 1 For the grins of it I replaced the T100P card with another newer card from inventory. This newer card has

Re: [Asterisk-Users] More words for Allison

On Sunday 11 January 2004 19:26, John Brown (CV) wrote: > How about Dollars Cents Euros Dollars, Cents, Zulu, and At are already done. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-user

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs

John, Pounds Sterling Screw You (purely optional :-) Your Call Is Being Connected Please Wait Sorry, Your Call Will Be Answered Soon Your Call Is Important. (That is why we have not yet bothered to answer it, instead dancing around the office high on tip-ex.) And for the girlfriend filter: Sorr

Re: [Asterisk-Users] More words for Allison

On Sunday 11 January 2004 18:36, John Todd wrote: > Here's the latest batch of words to get shipped out to Allison Smith. > Please submit reasonably small changes to me by tomorrow 10:00 AM > Eastern time, and I'll add them. > storm > warning > watch > thunderstorm > hail > weather > lightning > fo

Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision

Rich Adamson wrote: > > > Thanks to everyone that responded to my channel bank question. Ive > > decided that the Adit 600 would be a good choice. > > Then I got to thinking about SIP phones and wondered if their quality > > has progressed to the point that they can be deployed to customers who

[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2448 - 10 msgs

Some I've added, some already existed, some I declined. :-) JT Pounds Sterling Screw You (purely optional :-) Your Call Is Being Connected Please Wait Sorry, Your Call Will Be Answered Soon Your Call Is Important. (That is why we have not yet bothered to answer it, instead dancing around the o

[Asterisk-Users] sip and x-lite

Dear all, Can you give me the configurations for x-lite and sip in *. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.co

Re: [Asterisk-Users] More words for Allison

John Todd wrote: hurricane tornado You missed typhoon! Regards, Steve ___ 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/lis

Re: [Asterisk-Users] sip and x-lite

try this... http://www.fnords.org/~eric/asterisk/ cm - Original Message - From: "Ing Isianto Istiadi" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 12, 2004 7:50 AM Subject: [Asterisk-Users] sip and x-lite > > > Dear all, > Can you give me the configurations for

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

can u give me the configuration for the firewall?? with the same configuration i can't even talk or hear... its giving me the RTP Read Error whenever one picks up the phone. cm - Original Message - From: "Steve" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 12, 2004 6

Re: [Asterisk-Users] possible solution to PRI T100P dropped call issue

I have a T100P connected to an xspedius T1. We occasionally have calls disconnect. I just ran zttool and I get Tx/Rx Levels as 0/2. My zaptel.conf has --> span=1,0,0,esf,b8zs I can switch out the T100P and see what happens and report back. Walker On Sun, Jan 11, 2004 at 07:07:07PM -0700, Jo

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

Steve wrote: On Saturday 10 January 2004 06:07 pm, Owen Kelso wrote: I'm using Asterisk on a open server (no firewall or NAT) and trying to communicate with a Grandstream BudgeTone 102 SIP phone which is behind NAT. The BudgeTone is at firmware level 1.0.4.30 and Asterisk is from

Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

Thanks for all of your responses. I confirmed that the phone works perfectly without NAT or through a IPSec VPN (yeah, I know, same thing). I've concluded that the Netgear router (FVS318) performing the NAT is corrupting the outgoing RTP packets. Traces confirmed that the BudgeTone is sending th

[Asterisk-Users] questions

Dear all, I have activated call waiting (but since my pstn doesn't support call waiting, I can't test it with the pstn), and I have 3 fxses. But when I call the extentions (if that extention is already called), then I got the busy tones. Is it possible to use call waiting for fxs phone?

Re: [Asterisk-Users] More words for Allison

Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, January 11, 2004 7:36 PM Subject: [Asterisk-Users] More words for Allison > As usual, donations to what will be a ~$110 USD expense would be > appreciated, as I am paying for this round out o

Re: [Asterisk-Users] questions

- Original Message - From: "Ing Isianto Istiadi" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, January 11, 2004 11:30 PM Subject: [Asterisk-Users] questions > Dear all, > I have activated call waiting (but since my pstn doesn't support call > waiting, I can't test it with the

Re: [Asterisk-Users] More words for Allison

We sent $50 USD for the cause john brown chagres technologies On Mon, Jan 12, 2004 at 12:10:09AM -0500, Andrew Thompson wrote: > Original Message - > From: "John Todd" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Sunday, January 11, 2004 7:36 PM > Subject: [Asterisk-Users] More

Re: [Asterisk-Users] More words for Allison

Andrew Thompson wrote: Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, January 11, 2004 7:36 PM Subject: [Asterisk-Users] More words for Allison As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying

Re: [Asterisk-Users] Cisco 79xx Ringtones

Siggi Langauf wrote: I'm after two very specific ringtones for the 79xx's... If you want some classic office phone ringers: http://www.leonine.com/~lion/phones.php These are Merlin rings. For some, myself included, they're a bit nostalgic. Lion -- = lion is Lion J Templin

RE: [Asterisk-Users] More words for Allison

+$20 from us -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Sunday, January 11, 2004 10:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] More words for Allison Andrew Thompson wrote: > Original Message - > From: "Jo

RE: [Asterisk-Users] More words for Allison

I just sent $20.00 to [EMAIL PROTECTED] I am new to the list so I don't really know what I am donating to, but the whole Asterisk program sounds pretty cool and I hope to work myself to setting up an experimental system to play with it in the near future. Calvis Redmond, WA -Original Mess

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