Re: [Asterisk-Users] Someone please try MeetMe MOH with latest CVS and GS phone

2004-08-28 Thread Dave Cotton
On Fri, 2004-08-27 at 16:32 +, Tony Mountifield wrote: I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Brian McSpadden
Does this also effect 1.0-RC2? I am having a similar issue at a customer site over a frame relay network that is having occasional choppy sound over a fairly open line, with the jitter buffer enabled, as well as trunk=yes enabled. Thanks! Brian On Fri, 27 Aug 2004 12:47:05 -0700, Kris Boutilier

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Jeremy Bogan
http://www.dse.com.au/cgi-bin/dse.storefront/ 412ff6210573f994273fc0a87f9 c0726/Product/View/M9917 I emailed Dick Smith with the requirements but none of the power supplies they have can do the 1500mA. -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host

[Asterisk-Users] FXO probs in Aus. Should I give up?

2004-08-28 Thread Jamie Carl
Hey all, I've been trying to get my X101P working again as of late (it used to work great) and before I decide to trash the card I thought I'd post up my symptoms to see if anyone has any ideas. My old working config was basically 1 channel running fxsks signalling. It was working great with no

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Gary
the the real power requires and read the pages... like when can you belive any sales gimp who works for dse ?? its in their catalogue, you asked, goto a store and find it !! also check jaycar and altronics they are good bets as well. Gary On Sat, 28 Aug 2004 17:08:18 +1000, Jeremy Bogan

[Asterisk-Users] ISDN BRI card exepriences in UK

2004-08-28 Thread David Gurr
Looking for folks experiences with ISDN BRI cards in the UK ... what's good and what's bad and any gotchas. Thx -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Florian Overkamp
BTW, -Original Message- I emailed Dick Smith with the requirements but none of the power supplies they have can do the 1500mA. I've got an IAXy running on a 1100mA powersup I found packaged with my Lucent WLAN basestation. Be a little flexible and just try it. Only thing: If the

RE: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Roy Eddleston
Jeremy I am in the UK but the IAXy's (x5) I just received from the states have 120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated that this was what Digium are shipping with the IAXy. Digium state 1500mA on their docs but DigitNetworks state 1200mA on theirs, but ship the

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Jeremy Bogan
Hi Roy, I am in the UK but the IAXy's (x5) I just received from the states have 120VAC/9VDC 800mA supplies, I questioned this, but the supplier stated that this was what Digium are shipping with the IAXy. Thanks for the info. I bought a power supply that does 1500mA at 9V DC, but the IAXy doesn't

RE: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Roy Eddleston
Jeremy Apart from voltage and current the main thing to watch out for is that the PSU MUST be fully regulated or switched, an unregulated PSU will give you all sorts of weird problems. Before I opted for 240V to 120V mains converters (they were on offer) to use with the supplied US PSU's, I too

[Asterisk-Users] G729 licenses

2004-08-28 Thread Sergey Lapin
Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind

[Asterisk-Users] Disconnection From IAXTel

2004-08-28 Thread Sunrise Ltd
Muiz Motani wrote: I am using IAX soft clients [snip] on a NATed private LAN [snip] [snip] disconnected after 9 seconds. 8-10 seconds is roughly the time it takes for an IAX transfer to kick in. I have seen similar cases not with IAXtel but other IAX connections where end

Re: [Asterisk-Users] chan_sccp2 7960 -- documentation and examplerequest.

2004-08-28 Thread Craig Guy
I have now today also configured a 7960 to work with asterisk via chan_sccp. I have only used SCCP firmware 5.0 (5) with the 7960 and I gotta say that I much prefer the SIP 7.2 firmware. The real reason at this stage for going SCCP is for support of the 7914 expansion module. This phone will be

RE: [Asterisk-Users] AGI dtmf problems (with x-lite) (solved)

2004-08-28 Thread Raul Elizondo (wizardteam)
Hi Steven, IT is interesting you are even that far along with your AGI application when you haven't even figured out your mail client. Do you mean my email client? or my voicemail? Voicemail is working fine on digim extensions, i even changed the language to spanish (btw, there are lot of

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Jeremy Bogan
Apart from voltage and current the main thing to watch out for is that the PSU MUST be fully regulated or switched, an unregulated PSU will give you all sorts of weird problems. Maybe I have a busted IAXy, the power supply i've got is regulated and supports multiple voltages in different mA

[Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread Storm D. J. Petersen
Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect

[Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread Storm D. J. Petersen
Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect

Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk

2004-08-28 Thread gARetH baBB
On Mon, 23 Aug 2004, Robert Boardman wrote: Heartened by your that you have got x-lite working, I have been trying, but failing to now get x-lite working, don suppose you could send me a quick screen shot of you x-lite settings? Not really, but it's not hard to get going. Presuming you

Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

2004-08-28 Thread Tim Robinson
Tim Robinson wrote: Andrew, I am looking for exactly the same thing, except using the normal Dial command via Zaptel with the c option to defer answering til you press the # key. I am not a programmer but have briefly looked at the code in app_dial.c and I think it is doable - just need to

Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

2004-08-28 Thread Tim Robinson
Tim Robinson wrote: Andrew, I am looking for exactly the same thing, except using the normal Dial command via Zaptel with the c option to defer answering til you press the # key. I am not a programmer but have briefly looked at the code in app_dial.c and I think it is doable - just need to

[Asterisk-Users] switch statement in extensions.conf

2004-08-28 Thread Michael George
On the extensions.conf explanation page is a mention of the switch statement and it refers one to the connecting two * servers page. The only mention of the switch statement there is brief and in the example. However, the example seems to have some errors in it. It shows a sample of what's in

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Duane
Jeremy Bogan wrote: Maybe I have a busted IAXy, the power supply i've got is regulated and supports multiple voltages in different mA ratings, with the 9V DC at 1500mA. Have you tried feeding it less amps at all? -- Best regards, Duane http://www.cacert.org - Free Security Certificates

[Asterisk-Users] Re: Disconnection from IAXTel

2004-08-28 Thread Brad Ediger
Try an iax2 debug at the CLI and watch the logs as you get disconnected. See if it's something at the IAX protocol level. If you don't get any insight from that, use Ethereal on the segment to see what's going on. Brad I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Rich Adamson
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're using fairly current CVS code. There is something not right w/the trunking that causes choppy sound. See the wiki for more info. I am using current CVS code and I have trunk=no. Still sounds crappy. I need to

Re: [Asterisk-Users] FXO interfaces used in UK?

2004-08-28 Thread Rich Adamson
Impedance setting in the UK!? OK, I've clearly missed something along the way. A search on the Wiki says something about Zcomplex impedance but I have absolutely no idea where or what this is. If someone could point me in the right direction I'd be extremely grateful. That essentially

RE: [Asterisk-Users] G729 licenses

2004-08-28 Thread Kevin Walsh
Sergey Lapin [EMAIL PROTECTED] wrote: What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case?

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Scott Laird
On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of the time, systems

RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread steveb
All, I am experiencing this problem with an IAX link to the UK provider TelAppliant. chan_iax2.c:5251 socket_read: Rejected connect attempt from 217.14.132.162 Not sure what is causing this, however, it seems to have started sine I downloaded Asterisk CVS-HEAD-08/19/04-19:55:53. Not sure if

Re: [Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread Lyle Giese
What do you have in extensions.conf for inbound from FWD? In my default section, I have: exten = ${FWDNUMBER},1,Goto(housemenu,s,1) Lyle - Original Message - From: Storm D. J. Petersen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 09, 2100 10:09 AM Subject:

[Asterisk-Users] IAX dialing indication tone (PI = 8)

2004-08-28 Thread Matthew Oulton
Hi, I also have the same problem, from memory is this not progress indicator=8 thatdeals withthe dialing indicator ? Anyway I am also stuck with not having any dial indication, has anybody got an idea. MO ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread Bill Seddon
Just by way of giving you some encouragement, I too run a version downloaded on about the same day and use Telappliant successfully. I did get a rejected error until I converted one of my accounts to IAX from the default SIP. I don't think there is anything special about my setup and the * PBX

[Asterisk-Users] asterisks and vonage

2004-08-28 Thread Michael Di Martino
to start with i am new to asterisks and i am also a telcom idiot. with that said i have one vonage line i would like to hook up in my soon to be built Asterisk ippbx server. Now with the one Vonage (with call waiting) line can i receive more one call using an auto attendant route the call the

RE: [Asterisk-Users] Queue Announcement not until after # accept callpressed

2004-08-28 Thread Edward Eastman
This is something I'm after as well, what I have found is the following: http://bugs.digium.com/bug_view_page.php?bug_id=0001082 http://lists.digium.com/pipermail/asterisk-dev/2004-February/003201.html which pretty much does what I(you) want, the one problem with it is that while the agent is

[Asterisk-Users] UK Disconnect supervision with TDM400P

2004-08-28 Thread Edward Eastman
Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK.

[Asterisk-Users] SIP Provider for Reseller

2004-08-28 Thread Beierlein Moritz
Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] incomming call rejected using IAX2 with FWD

2004-08-28 Thread Sunrise Ltd
Storm D. J. Petersen wrote: I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message:

Re: [Asterisk-Users] Queue Announcement not until after # accept call pressed

2004-08-28 Thread Greg Boehnlein
On Fri, 27 Aug 2004, Andrew Brown wrote: When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after

[Asterisk-Users] Broadvoice problem

2004-08-28 Thread Russell Horn
Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '[EMAIL

RE: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Marty Mastera
Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for

Re: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Ed Brady
I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses for broadvoice's IP servers, so I had to change the

Re: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Ed Brady
One other thing that I forgot to mention, while I was hunting this problem down. Broadvoice also reprovisioned my account, which caused me to have to get a new password. I do not know if you will need to do this or not. Ed Ed Brady wrote: I had the same problem. To fix it, I had to do two

RE: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Marty Mastera
I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses for broadvoice's IP servers, so I had to

RE: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Marty Mastera
I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses for broadvoice's IP servers, so I had to

Re: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Ed Brady
Marty Mastera wrote: I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses

Re: [Asterisk-Users] Broadvoice problem

2004-08-28 Thread Ed Brady
Marty Mastera wrote: I had the same problem. To fix it, I had to do two things First: I had to update to CVS head, this was as per broadvoice support. Second: After updating, I had to change my sip.conf. Originally my sip.conf used hard coded ip addresses

[Asterisk-Users] POE

2004-08-28 Thread Steve Maroney
Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

Re: [Asterisk-Users] POE

2004-08-28 Thread Michael Welter
Steve Maroney wrote: Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Michael George
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to auto-negotiate their half vs full duplex nic interfaces. About 50% of

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Michael George
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote: On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into routers, switches, servers, etc, that are allowed to

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Michael Graves
On Sat, 28 Aug 2004 15:24:01 -0400, Michael George wrote: On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote: On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote: I do a lot of work with companies throughout the US on network performance and we _frequently_ run into

Re: [Asterisk-Users] UK Disconnect supervision with TDM400P

2004-08-28 Thread Richard Scobie
Edward Eastman wrote: I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the remote party hangs up, I get approx 20secs of the bt line hungup tone before

Re: [Asterisk-Users] POE

2004-08-28 Thread Greg Boehnlein
On Sat, 28 Aug 2004, Michael Welter wrote: Since the Cisco 79XX phones preceded the PoE standard, they are different--polarity is reversed. IANAE, but as I understand the PoE devices, there are two types--one always applies -48VDC to the brown pair while the other senses (as per the PoE

[Asterisk-Users] Broadvoice BYOD Plans

2004-08-28 Thread Ben Wern
Can anyone who is ___ 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

[Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting

2004-08-28 Thread Ben Wern
Can anyone who is using Asterisk with Broadvoice tell of their experiences with 3-way calling and call waiting? I can't get Broadvoice to respond to my question, but I understand that there is a per minute fee (3.9 c/minute?) if you go over your use allowances. My question is, how are 3 way

Re: [Asterisk-Users] POE

2004-08-28 Thread Michael Welter
Greg Boehnlein wrote: On Sat, 28 Aug 2004, Michael Welter wrote: Since the Cisco 79XX phones preceded the PoE standard, they are different--polarity is reversed. IANAE, but as I understand the PoE devices, there are two types--one always applies -48VDC to the brown pair while the other senses

Re: [Asterisk-Users] Disconnection From IAXTel

2004-08-28 Thread Muiz Motani
How do I go about disallowing transfers when I am running an IAX soft phone. Is that setting not at the * server? Obviously, I don't have control over the configuration of the IAXTel * server. On 28 Aug 2004 at 19:03, you wrote: Disallowing an IAX transfer has always solved these problems

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Andrew Kohlsmith
On Saturday 28 August 2004 15:00, Michael George wrote: The difference between that and what I'm getting from IAX/GSM is profound, with GSM being intolerably poor quality. That's odd; every single voice call coming in and out of the company I work for is using the GSM codec with asterisk and

Re: [Asterisk-Users] IAXy Power in Australia?

2004-08-28 Thread Jeremy Bogan
Have you tried feeding it less amps at all? Not yet, but i'll see if I can find a power supply with less amps. -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Andrew Kohlsmith
On Saturday 28 August 2004 15:24, Michael George wrote: I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause when I use Zap -- Zap or Zap -- SIP there is no en/decoding

Re: [Asterisk-Users] POE

2004-08-28 Thread Greg Boehnlein
On Sat, 28 Aug 2004, Michael Welter wrote: Greg Boehnlein wrote: On Sat, 28 Aug 2004, Michael Welter wrote: Since the Cisco 79XX phones preceded the PoE standard, they are different--polarity is reversed. IANAE, but as I understand the PoE devices, there are two types--one always

Re: [Asterisk-Users] SIP Provider for Reseller

2004-08-28 Thread Roger Schreiter
Beierlein Moritz schrieb: Hi List, does somebody know a SIP Provider which offers reseller possibilities? ... Hi, you are from Germany? Sorry for advertising: http://voip.planinternet.net We are VoIP carrier and are supporting several German VoIP providers. Please contact us directly, if you

[Asterisk-Users] Newbie

2004-08-28 Thread Michael Di Martino
I am interested in setting up an Asterisk server as my home phone system. I ultimately want one 10 digit phone number, three extensions, and an auto attendant My current phone service provider is Vonage, I have one line with call waiting. My concern is will I need to add additional lines if

Re: [Asterisk-Users] SIP Provider for Reseller

2004-08-28 Thread sgup015
VoiceValley will do soon in NZ. Quoting Beierlein Moritz [EMAIL PROTECTED]: Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Newbie

2004-08-28 Thread Greg Broiles
There are at least two ways to configure Vonage and Asterisk to connect to each other: you can connect the FXS port on your Vonaga ATA to an FXO port on an Asterisk box, or you can make a SIP connection to Vonage's server that Vonage thinks is coming from a soft phone. If you connect the ports

Re: [Asterisk-Users] Newbie

2004-08-28 Thread Michael Graves
On Sat, 28 Aug 2004 17:44:53 -0700, Greg Broiles wrote: There are at least two ways to configure Vonage and Asterisk to connect to each other: you can connect the FXS port on your Vonaga ATA to an FXO port on an Asterisk box, or you can make a SIP connection to Vonage's server that Vonage thinks

RE: [Asterisk-Users] SIP Provider for Reseller

2004-08-28 Thread Chad Brown
I've been in discussions with broadvoxdirect. [EMAIL PROTECTED] has been encouraging our company to sign up. I have yet to do so. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, August 28, 2004 3:43 PM To: Asterisk

Re: [Asterisk-Users] POE

2004-08-28 Thread Craig Guy
We are using a Netgear FSM7326P to PoE a 7960 (with 7914 attached). Craig - Original Message - From: Michael Welter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, August 29, 2004 2:13 AM Subject: Re: [Asterisk-Users] POE

RE: [Asterisk-Users] asterisks and vonage

2004-08-28 Thread Paterson, Mark
Well, so I'm unable to get any inbound calls from Vonage but can call outbound all day long. If you are able to get inbound calls working I would appreciate it if you could share your configs via personal email. Rgs, [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED]

Re: [Asterisk-Users] POE

2004-08-28 Thread Kevin Day
On Aug 28, 2004, at 1:13 PM, Michael Welter wrote: Since the Cisco 79XX phones preceded the PoE standard, they are different--polarity is reversed. IANAE, but as I understand the PoE devices, there are two types--one always applies -48VDC to the brown pair while the other senses (as per the

RE: [Asterisk-Users] Are there any graphic designers on this list?

2004-08-28 Thread Paterson, Mark
Is there an Asterisk Assistant for linux or windows? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw Sent: Friday, August 27, 2004 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Are there

Re: [Asterisk-Users] POE

2004-08-28 Thread Steven Pritchard
On Sat, Aug 28, 2004 at 12:57:48PM -0500, Steve Maroney wrote: I was wondering what POE solutions are being used ? The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working fine for me with my snom 200 phones. Steve -- Steven Pritchard - KS Pritchard Enterprises, Inc. Email: [EMAIL

[Asterisk-Users] Distinctive ring detection problem

2004-08-28 Thread Paul Budden
I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Michael George
On Sat, Aug 28, 2004 at 05:08:30PM -0400, Andrew Kohlsmith wrote: On Saturday 28 August 2004 15:24, Michael George wrote: I just saw a page on the wiki that mentions that running X11 or a VESA frame buffer can cause jittery sound. I only have this problem with IAX2, but that might be cause

Re: [Asterisk-Users] POE

2004-08-28 Thread Kevin P. Fleming
Steven Pritchard wrote: The dumb, cheap ($30 retail) 3com PoE injectors (3CNJPSE) are working fine for me with my snom 200 phones. And there are plenty of places to buy them cheaper as well. Provantage has them for around $22. www.soekris.com (a maker of embedded PC systems) has the same unit

Re: [Asterisk-Users] iaxtel and jitterbuffer

2004-08-28 Thread Andrew Kohlsmith
On Saturday 28 August 2004 23:01, Michael George wrote: It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not running and the Framebuffer has been turned off in /boot/grum/menu.lst. I have disabled all the servers except for sshd. I have the latest source from CVS HEAD as of

Re: [Asterisk-Users] IAX dialing indication tone (PI = 8)

2004-08-28 Thread spkao
I played around with extensions.conf and found that if you just use the 'r' option in Dial app and leave out the other options t, T, m, ... then you get dial indication. Haven't got around to try out the working option combinations but there it is. PK - Original Message -

Re: [Asterisk-Users] Disconnection From IAXTel

2004-08-28 Thread Dave Cotton
On Sat, 2004-08-28 at 14:01 -0700, Muiz Motani wrote: How do I go about disallowing transfers when I am running an IAX soft phone. Is that setting not at the * server? Obviously, I don't have control over the configuration of the IAXTel * server. Which softphone are you using? -- Dave

[Asterisk-Users] where can I find spandsp?

2004-08-28 Thread Rich Adamson
Seems the opencall.org site has basically been unavailable for days/weeks. Is there another location to obtain the current code? Also, will spandsp install against the current * cvs? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED]