RE: [Asterisk-Users] Where's the Fiber
We have an unframed E1 used for data, and it is fiber all the way to our server room, and then broken out to a G.703 interface. A few of the E1's I've seen lately for voice have actually been g.shdsl to the premises with an interface converter between that and the pbx. You can always rely on your telco to do what they need to do for the minimum investment possible, and if that means using fiber instead of copper or visa versa then that's what they'll do. As long as the media can support the bandwidth it almost doesn't matter what goes on inbetween, as long as it comes out in the right format (eg T1 in your case) at each end. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, 18 June 2006 09:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Where's the Fiber Where's the Fiber? I was reading about T1 lines and came across this statement.. It basically said T1's are made up of copper...Wasn't T1 made up of Fiber? Is the new trend to move T1 away from fiber and use copper? http://www.pulsewan.com/data101/pdfs/t1basics.pdf#search='t1%20via%20cop pe r' page 4 T1 Physical Characteristics A T1 is physically made up of two balanced pairs of copper wire (commonly known as twisted pair). The pairs are used in a full duplex configuration where one pair transmits information and the other pair receives information. Customer Premises Equipment (CPE) typically terminate a T1 with a RJ-48C jack. The following illustration shows a typical T1 cable and interface. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, June 17, 2006 5:08 AM Subject: Re: [Asterisk-Users] T1 Copper or T1 Fiber Line Thanks for the inso... So T1 lines in the United States also use copper lines from the company to the telephone exchange in some installations? What's the benefit to the subscriber to this? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 16, 2006 9:11 AM Subject: Re: [Asterisk-Users] T1 Copper or T1 Fiber Line Any T1 I've seen in the last 3 years has actually been DS1-over-HDSL2. What comes in to the building is a single pair of copper into the smartjack, and then you have a traditional DSX1 to plug in to. I don't think real T1s (in the physical sense) have existed for years. Before DS1-over-HDSL2 the ones I had provisioned were DS1-over-HDSL (2 copper pairs)... never had a real, genuine T1. But again... you don't get to play with that side of it. You order a T1, you get a smartjack that has a T1 jack (DSX1) on it and what's on the other side is irrelevant. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail with NFS
On Sat, Jun 17, 2006 at 09:31:54PM -0500, Aaron Daniel wrote: On Sat, 17 Jun 2006, Douglas Garstang wrote: Other applications can handle it. Don't see why Asterisk can't. Mount the nfs volume with the -soft option. Do a 'df -k' and you will see that the df command will time out in a couple of seconds. Why can't Asterisk do the same? Just gonna throw gas on the fire. df -h doesn't continuously poll the drive for data, asterisk is (for mwi). So each timeout turns into another timeout. Didn't you already test changing the time on checkmwi? And did it not change the behavior (not necessarily for the better)? Your theory is easy to check with watching the state of the main asterisk thread, or maybe strace -p MAIN_ASTERISK_PID . If it remains hung constantly in a D state on the same system call, a shorter timeout just wouldn't have helped. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which application to open Zap channel?
I'm using Dial(Zap/X/) as suggested. However, Dial(Zap/X) does indeed work for me. So I'm curious, what's the difference between them, and when wouldn't just Zap/X work? On Wed, 2006-06-14 at 11:14 -0500, Eric ManxPower Wieling wrote: Carey O'Shea wrote: I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI NetJet
I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? There is an mISDN driver available on sourceforge: http://sourceforge.net/projects/misdn4oz It works pretty well under chan_capi (using the misdn capi driver), but doesn't work with the latest (mqueue) branch of mISDN. The maintainer is working on porting it though. I have his work-in-progress code and it is pretty broken - I can make a call but it crashes on call end. I was actually looking for the bugs today but turned off the watchdog on my test server by mistake so it will remain crashed until I get into the office tomorrow :( James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ISDN BRI NetJet
On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote: I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 It could work with the deprecated chan_modem. Don't wast your time. Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? Replace it by some Cologne Chip Card. The single port card is cheap. Than you can use bristuff, chan_misdn or visdn. Assuming you can get such a card where you live. In Australia there is no PCI HFC adapter available. We can get the AVM fritz card, but the importers have marked the price up quite substantially based on the fact that they have A-Tick approval on it. But as per my last post, the netjet mISDN drvier doesn't currently work, so if you can find an alternate card where you are then do that. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold troubleshooting
I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to install this rpm package. But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3 sound and song etc. What are your views?? Regards, Amna Saleem On 6/17/06, Sharon Lim [EMAIL PROTECTED] wrote: Did you install the sound packages such as mpg123-0.59r-1.i386.rpm ? Can download from http://rpm.pbone.net/index.php3/stat/4/idpl/516450/com/mpg123-0.59r-1.i386.rpm.html good luck! On 6/16/06, kharris [EMAIL PROTECTED] wrote: Can anyone point me in the direction for resources for troubleshootingno MusicOnHold with Asterisk version 1.2.9.1 and Asterisk Addons version1.2.3?ThanksKarl___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free sun boxes
I'm in southern California, are you close or can you ship? Bob Knight wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold troubleshooting
On Sun, Jun 18, 2006 at 11:50:12AM +0500, amna saleem wrote: I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to install this rpm package. But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3 sound and song etc. What are your views?? Regards, Amna Saleem Unless you need to stream mp3 music, playing an mp3 music file will be a waste of CPU. mp3 files are highly compressed, but need to be downsampled to 8khz mono (16 bit samples). If you'll convert the mp3 file to wav and downsample, chances are you'll end up with a comparable disk space and much less work for playing. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold troubleshooting
So the problem still persisits. What should I do? My musiconhold is not playing :) On 6/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Jun 18, 2006 at 11:50:12AM +0500, amna saleem wrote: I have read in wiki pages that for astreisk 1.2.9.1 , you don`t have to install this rpm package. But Ialso read that Red hat Linux 9 and enterprise doesn`t suppport mp3 sound and song etc. What are your views?? Regards, Amna SaleemUnless you need to stream mp3 music, playing an mp3 music file will be awaste of CPU.mp3 files are highly compressed, but need to be downsampled to 8khz mono(16 bit samples). If you'll convert the mp3 file to wav and downsample, chances are you'll end up with a comparable disk space and much lesswork for playing.--Tzafrir Cohensip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406[EMAIL PROTECTED]http://www.xorcom.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What ever happened to the LTAPI, the Linux Telephony API?
Hi, I've just been going through the various modules that are autoloaded to see what I need and what I don't and came across chan_phone.so which loads /etc/asterisk/phone.conf. I did a lookup on voip-info and google and came across this article in Linux Journal from 2001. Anyone know why it isn't being used much (from what I can tell) and what's happening with it today? Thanks, Mike http://www.linuxjournal.com/article/4468 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ISDN BRI NetJet
For a single BRI the Netjet is the way to go. I have the card running with the mISDN drivers. I haven't tried capi yet but seems to work great in ptp mode. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: Sunday, 18 June 2006 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: ISDN BRI NetJet On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote: I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 It could work with the deprecated chan_modem. Don't wast your time. Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? Replace it by some Cologne Chip Card. The single port card is cheap. Than you can use bristuff, chan_misdn or visdn. Assuming you can get such a card where you live. In Australia there is no PCI HFC adapter available. We can get the AVM fritz card, but the importers have marked the price up quite substantially based on the fact that they have A-Tick approval on it. But as per my last post, the netjet mISDN drvier doesn't currently work, so if you can find an alternate card where you are then do that. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon Berlin Starts Tomorrow (Montag)
If you're in or near Berlin and want to join in the fun, head to the Estrel Hotel tomorrow (Monday / Montag). Tickets will be available at the door. Check-in opens at 8:00 AM. We're expecting a great time for all! Image of the early arrivals geeking out: http://www.sokol-associates.com/files/images/astricon1.jpg Hope to see you here! Thanks, Steve -- Steven Sokol AstriCon 2006: http://www.astricon.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 delivered via Copper
If my Telco tell me that they can give me a T1 delivered via Copper how many options does the Telco company have. Option 1) T1 is carried over some form of DSL Option 2) What is the ??? Anyone has any idea what option 2 is? Thanks --Davi-Ann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 delivered via Copper
On Sunday 18 June 2006 14:07, [EMAIL PROTECTED] wrote: If my Telco tell me that they can give me a T1 delivered via Copper how many options does the Telco company have. That's up to them. You could ask them what delivery options they have for T1. Option 1) T1 is carried over some form of DSL Have no experience with T1, but with E1, as someone mentioned earlier on this list, when carried over copper, an HDSL carrier is not uncommon, with an HDSL modem at your end and another at the telco's end. Some of these HDSL modems can do between 5.8Km to 10Km. However, if the telco can deliver the T1 over a fibre up to your building's telecomms infrastructure, or to your own facility assuming they can terminate their fibre onto some node there, you wouldn't have to worry too much about that. Mark. pgplEmKQStl5b.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free sun boxes
how much are you selling that stuff? On 6/18/06, Mike Fedyk [EMAIL PROTECTED] wrote: I'm in southern California, are you close or can you ship? Bob Knight wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lito Manansala www.voicefidelity.net Mobile: +63 906 437 0459 DID: (+63) 44 7906292 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 302 Redirecting support
Hello, I have a question . dose asterisk supports 302 Redirecting... ? I have SIP Server Not Asterisk and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output - -- Got SIP response 302 Redirecting... back from SIP SERVER IP -- Now forwarding SIP/108-ce60 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/67888-91de) Jun 18 16:47:19 NOTICE[11251]: chan_local.c:378 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Jun 18 16:47:19 NOTICE[11251]: app_dial.c:232 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' == Everyone is busy/congested at this time -- Executing Hangup(SIP/108-ce60, ) in new stack == Spawn extension (internal, 420026, 3) exited non-zero on 'SIP/108-ce60' -- afte i googled a little and i find this http://lists.digium.com/pipermail/asterisk-users/2006-April/146983.html .. so dose this means that the Redirecting is not supported ? I'm using a little bit old Asterisk 1.0.x but i think it should work ? Thank You Best Regards Sherif Nagy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 302 Redirecting support
On 6/18/06, Sherif Nagy [EMAIL PROTECTED] wrote: Hello, I have a question . dose asterisk supports 302 Redirecting... ? I have SIP Server Not Asterisk and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output - -- Got SIP response 302 Redirecting... back from SIP SERVER IP -- Now forwarding SIP/108-ce60 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/67888-91de) Jun 18 16:47:19 NOTICE[11251]: chan_local.c:378 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Jun 18 16:47:19 NOTICE[11251]: app_dial.c:232 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' == Everyone is busy/congested at this time -- Executing Hangup(SIP/108-ce60, ) in new stack == Spawn extension (internal, 420026, 3) exited non-zero on 'SIP/108-ce60' -- Redirecting is supported, but only to peers that are registered directly with the Asterisk server. Asterisk will not pass the 302 back to the calling UA because Asterisk isn't a SIP proxy, it's a B2BUA. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which application to open Zap channel?
I don't know for sure, but I can make a few guesses. Dial(Zap/X) would be used if you are dialing an FXS port, since these ports don't normally require digits to be sent. Dial(Zap/X/) would be used if you are dialing an FXO port, since these ports almost always require digits to be dialed. In this example you are dialing no digits. Carey O'Shea wrote: I'm using Dial(Zap/X/) as suggested. However, Dial(Zap/X) does indeed work for me. So I'm curious, what's the difference between them, and when wouldn't just Zap/X work? On Wed, 2006-06-14 at 11:14 -0500, Eric ManxPower Wieling wrote: Carey O'Shea wrote: I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which phones are good, or at least acceptable, for home and office
OK, but I felt that I had narrowed the field down a bit and was just looking for confirmation that there were no severe problems with the choices. People like yourself obviously have some experience with the different phone models and that is what I was hoping to learn from. Yes, I could go buying all sorts of phones to find ones that are acceptable but my life is not telephony, I just want it to work. A Porche certainly does the job better than a Chevette but if the Chevette is adequate for the job why not use it. I think that the Polycom 601 is nice but I don't see that it provides anything that I need beyond what the 501 gives and is certainly overkill for the satellite phones. Mike Michael Graves wrote: I can't tell you how many times I've seen broad questions like this posted to the list.. The wiki (www.voip-info.org) is your friend. Use it. There's a lot of good advise there. Google is also your friend. Use it, too. Most especially use it to search the list archives. There was just a long thread about this a few days ago. Finally, you can do what I did...buy some phones, try them for a while then resell the ones you don't like. Ebay is a great toolo for this. I bought and sold eight different model of SIP phones before settling upon what I use today. When you've gained enough experience to have some well founded opinions add to the wiki. Lastly, if you're going to buy serious desk phones try the Aastra 480i CT and the Polycom IP600/601. Life's too short to use a cheap phone. Michael On Sat, 17 Jun 2006 20:35:02 -0400, M.Hockings wrote: I am looking to replace all of the old Bell (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable, headset capable) 3. primary office phone (good quality, headset, speaker phone) In most places the LAN wiring is already in place so the phone would need to be able to provide a LAN port for an existing computer. POE would be desirable in a couple of places due to limited power outlets. What I have considered is the Grandstream BudgeTone BT-102 or BT-200 for the satellite phones, a Grandstream GXP-2000 for the domestic phone as it has all the requirements and there is a POE device available for it. My alternative pick for this would be a Polycom 301. And for my office I was considering a Polycom 501. Are any of these choices known to be bad performers, hard to configure with Asterisk, etc. I have read that it is difficult or not possible to get the message waiting indicator to show for the BT-102. Is this a problem with the GXP-2000 or Polycom phones ? Also is it possible to use the Linksys POE injector/splitter to power a BT-102 ? Or are there other solutions for POE? Some Web references follow for the keen. Thanks for any thoughts or input on this. Mike Linksys POE Injector/Splitter _http://www.insight.ca/apps/productpresentation/index.php?format=printproduct_id=LNKPPOE12_ BT-102 _http://www.canadianvoipstore.com/product_info.php?cPath=95_105products_id=40_ GXP-2000 _http://www.canadianvoipstore.com/product_info.php?cPath=95_106products_id=331_ Polycom 301 _http://www.canadianvoipstore.com/product_info.php?products_id=757_ Polycom 501 _http://www.canadianvoipstore.com/product_info.php?products_id=758_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: _http://lists.digium.com/mailman/listinfo/asterisk-users_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Which phones are good, or at least acceptable, for home and office
On Sun, 18 Jun 2006 11:03:42 -0400, M.Hockings wrote: OK, but I felt that I had narrowed the field down a bit and was just looking for confirmation that there were no severe problems with the choices. People like yourself obviously have some experience with the different phone models and that is what I was hoping to learn from. Yes, I could go buying all sorts of phones to find ones that are acceptable but my life is not telephony, I just want it to work. Then buy something from a vendor that will take on the headaches for you. Any form of Asterisk implies quite a learning curve. You have to be willing to climb that in one way or another. A Porche certainly does the job better than a Chevette but if the Chevette is adequate for the job why not use it. I think that the Polycom 601 is nice but I don't see that it provides anything that I need beyond what the 501 gives and is certainly overkill for the satellite phones. It certainly depends upon what you're after...and what you feel is important. I don't like the low resolution LCD display on the lesser Polycom models. When I saw the backlit display on the Aastra 480 that was enough to compel me to buy one. Michael Mike Michael Graves wrote: I can't tell you how many times I've seen broad questions like this posted to the list.. The wiki (www.voip-info.org) is your friend. Use it. There's a lot of good advise there. Google is also your friend. Use it, too. Most especially use it to search the list archives. There was just a long thread about this a few days ago. Finally, you can do what I did...buy some phones, try them for a while then resell the ones you don't like. Ebay is a great toolo for this. I bought and sold eight different model of SIP phones before settling upon what I use today. When you've gained enough experience to have some well founded opinions add to the wiki. Lastly, if you're going to buy serious desk phones try the Aastra 480i CT and the Polycom IP600/601. Life's too short to use a cheap phone. Michael On Sat, 17 Jun 2006 20:35:02 -0400, M.Hockings wrote: I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable, headset capable) 3. primary office phone (good quality, headset, speaker phone) In most places the LAN wiring is already in place so the phone would need to be able to provide a LAN port for an existing computer. POE would be desirable in a couple of places due to limited power outlets. What I have considered is the Grandstream BudgeTone BT-102 or BT-200 for the satellite phones, a Grandstream GXP-2000 for the domestic phone as it has all the requirements and there is a POE device available for it. My alternative pick for this would be a Polycom 301. And for my office I was considering a Polycom 501. Are any of these choices known to be bad performers, hard to configure with Asterisk, etc. I have read that it is difficult or not possible to get the message waiting indicator to show for the BT-102. Is this a problem with the GXP-2000 or Polycom phones ? Also is it possible to use the Linksys POE injector/splitter to power a BT-102 ? Or are there other solutions for POE? Some Web references follow for the keen. Thanks for any thoughts or input on this. Mike Linksys POE Injector/Splitter _http://www.insight.ca/apps/productpresentation/index.php?format=print_id=LNKPPOE12_ BT-102 _http://www.canadianvoipstore.com/product_info.php?cPath=95_105_id=40_ GXP-2000 _http://www.canadianvoipstore.com/product_info.php?cPath=95_106_id=331_ Polycom 301 _http://www.canadianvoipstore.com/product_info.php?products_id=757_ Polycom 501 _http://www.canadianvoipstore.com/product_info.php?products_id=758_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: _http://lists.digium.com/mailman/listinfo/asterisk-users_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.pFrom: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten =
_40002,1,Dial(SIP/40002,30) From: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3
40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels
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Re: [Asterisk-Users] What ever happened to the LTAPI, the Linux Telephony API?
Mike Fedyk wrote: Hi, I've just been going through the various modules that are autoloaded to see what I need and what I don't and came across chan_phone.so which loads /etc/asterisk/phone.conf. I did a lookup on voip-info and google and came across this article in Linux Journal from 2001. Anyone know why it isn't being used much (from what I can tell) and what's happening with it today? The hardware is pretty crappy. I've got three of the linejack cards sitting on a shelf collecting dust. IIRC there are manifold limitations in terms of function and quality, and I think development has stopped because there are so many better hardware solutions on the market nowadays. MO, FWIW. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi, STREAM FILE and SIGHUP
Hi, I have developed a custom AGI in C++. Whenever I stream a file or say out digits with STREAM FILE and SAY NUMBER and hangup the call in between the AGI ends abruptly. I did a bit of surfing through previous posts and found out that asterisk sends a SIGHUP signal as soon as a caller ends a call. The suggesion was to catch the SIGHUP signal in the process and ignore it. I wrote the following piece of code at the star of the agi. #include csignal struct sigaction my_action; my_action.sa_handler = SIG_IGN; my_action.sa_flags = SA_RESTART; int sigret = sigaction (SIGHUP, my_action, NULL); This should solve the problem but unfortunately the agi is still crashing straight after I hangup the call. May be I need to unblock the signal, I am not sure how. Am I missing something. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail with NFS
On Saturday 17 June 2006 01:55, Tzafrir Cohen wrote: On Fri, Jun 16, 2006 at 09:40:35AM -0600, Mike Diehl wrote: I don't know how big your voicemail system is, but have you considered using Unison to syncronize the vm accross all your servers? I'm deploying multiple servers with two vm servers, each sync'ed every 5? minutes. If one fails, the other one should be good enough. The voicemail code assumes some locking semantics supported by the filesystem (sysv locks?) That's a relief. I always hope programs are smart enough to lock files they change. What happens when you sync a locked file? Then the lockfile will be created and deleted on the backup server as well. Since we kind of assume a master/slave situation with the two vm systems, I think this should work? But talk is cheap and I've not done this, yet. On the other hand, you might use the external notification option in voicemailmain to initiate the sync Isnt there a problem with samba with the very same issue? Not sure. I only minimally support samba these days. Hope this helps, Mike. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] agi, STREAM FILE and SIGHUP
I have developed a custom AGI in C++. Whenever I stream a file or say out digits with STREAM FILE and SAY NUMBER and hangup the call in between the AGI ends abruptly. I did a bit of surfing through previous posts and found out that asterisk sends a SIGHUP signal as soon as a caller ends a call. The suggesion was to catch the SIGHUP signal in the process and ignore it. I wrote the following piece of code at the star of the agi. You might want to read this: http://bugs.digium.com/view.php?id=6491 -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 delivered via Copper
It is not up to you anyhow, the local phone company that gives the last mile decides that. Usually it get decided on how far away you are from the CO. In any case they will test it before they tell you it's up and running to make sure it's a clean line, with a low level of noise. The testing goes for Fiber as well. Just like you have no control if the TDM network you use when making a call across country is going over Fiber and on top of that ATM SONET or even Satellite, you have no control (nor should you have) how they bring the T1 to your smartjack. Fiber is just a medium by which the upper level (layer) protocols are transmitted. it happens to be that fiber lasts longer than copper, and can take much more than copper, but if bandwidth and/or distance is not an issue, copper is just as good. On 6/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If my Telco tell me that they can give me a T1 delivered via Copper how many options does the Telco company have. Option 1) T1 is carried over some form of DSL Option 2) What is the ??? Anyone has any idea what option 2 is? Thanks --Davi-Ann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?
Hello,Long time subscriber/reader of this list - thank you for all the great ideas.Scenario:We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality:- Agents having the ability to login to a specific queue- Call distributed to that queue based on criteria- Basic reporting (ASA, AHT etc..) Solutions:- Flip the Mitel phones to load a SIP firmware and speak to AST (althought i'd love it, the powers that be probably won't)- Use the Asterisk queueing ability to send calls off network (AST) to the NCI platform (the Asterisk box can send these calls via SIP or TDM through a gateway). Goals:I'd like to create an Asterisk server running multiple queues for multiple tenants (or customers) that can provide the ability for agents to login remotely (either via an ingress call to AST or a www gui). The call flow would be similar to this: Agent#1 - logs into Mitel phoneAgent#1 - Dials XXX XXX into AstersikAgent#1 - Hears a prompt on Asterisk to login to a specifc queueAgent#1 - Passes DTMF and becomes 'available' in the eyes of Asterisk Agent#1 - Is now in queue*repeat for three agents*Now, all three agents are in an available state to Asterisk, and logged into our one queue. If Asterisk receives a call on a specific DID it will attempt to send the goal to agent#1, if agent#1 rings three times or returns a 'busy here' the call will pass to agent#2 etc. The challenge I see will be configuring an off-network queue, is anyone working with a similar setup?Does anyone have any thoughts on how to better accomplish my goals?Thanks in advance./Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice - Last Straw!
I guess its good I left them about two months after I joined them :)Robert Mann [EMAIL PROTECTED] wrote: I have always been an advocate for Broadvoice. Their service although is a little shoty at times has been an extremely cheap service that works with Asterisk. 19.99 a month for unlimited (Some say now really unlimited but I average 3500 minutes a month so pretty fine for me) calling and to several different countries. But I am on my last straw with them. The latest is somehow my primary and secondary number (as well as everyone else in the 661 area code using them) got their numbers LNP'ed (Local Number Porting) over to a terminal server (Dialup modem server) for Option 1 communications (O1 communications) about a week ago and there is still no ETA to get fixed. So when anyone calls our numbers they get a modem. I call them daily and get the we are sorry but we have no ETR yet. This is just plain crazy. So with all that does anyone know of another provider that offers unlimited or at least some sort of bulk minute deal that I can switch over to and get off this service and have real service?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Talk off
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse or maybe my patience
is thinning. All my calls go out and come in pstn through an FXO as I do not
have high speed available here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another location(that has
high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than yes
or no, so I looked in chan_zap.c and found relaxdtmf in many places but it
looked to my inexperienced eye that it could only be set to 'yes' or 'no',
but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is
indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
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Fwd: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?
-- Forwarded message --From: Christopher Aloi [EMAIL PROTECTED]Date: Jun 18, 2006 9:52 PM Subject: Re: FW: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts?To: Alexander Lopez [EMAIL PROTECTED]Alexander,Thanks for your reply, may I ask a few questions?- Does the Asterisk server maintain any type or presence for the agents? (i'm assuming this wouldn't be possible since your shooting the call out POTS) - How do your off-network callback agents identify their location to the Asterisk server?- Are you able to describer your dialplan configuration in detail?Thanks again,/Chris On 6/18/06, Alexander Lopez [EMAIL PROTECTED] wrote: I do this type of thing right now, with both agents that are logged in and callback agents, All off site and via PSTN From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christopher Aloi Sent: Sunday, June 18, 2006 8:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Creating Queues on Asterisk server - Sendingingress calls off-net to either PSTN or another VoIPapplication - thoughts? Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to login to a specific queue - Call distributed to that queue based on criteria - Basic reporting (ASA, AHT etc..) Solutions: - Flip the Mitel phones to load a SIP firmware and speak to AST (althought i'd love it, the powers that be probably won't) - Use the Asterisk queueing ability to send calls off network (AST) to the NCI platform (the Asterisk box can send these calls via SIP or TDM through a gateway). Goals: I'd like to create an Asterisk server running multiple queues for multiple tenants (or customers) that can provide the ability for agents to login remotely (either via an ingress call to AST or a www gui). The call flow would be similar to this: Agent#1 - logs into Mitel phone Agent#1 - Dials XXX XXX into Astersik Agent#1 - Hears a prompt on Asterisk to login to a specifc queue Agent#1 - Passes DTMF and becomes 'available' in the eyes of Asterisk Agent#1 - Is now in queue *repeat for three agents* Now, all three agents are in an available state to Asterisk, and logged into our one queue. If Asterisk receives a call on a specific DID it will attempt to send the goal to agent#1, if agent#1 rings three times or returns a 'busy here' the call will pass to agent#2 etc. The challenge I see will be configuring an off-network queue, is anyone working with a similar setup? Does anyone have any thoughts on how to better accomplish my goals? Thanks in advance. /Chris ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple port
Hi, Does asterisk support mutl-port binding? Say beside setting the port 5060 in sip.conf, I want to use another port, say 6060. How can I set to use more than one port. Is it possible? unplug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where's the Fiber
Make fibre your friend. -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 James Harper wrote: We have an unframed E1 used for data, and it is fiber all the way to our server room, and then broken out to a G.703 interface. A few of the E1's I've seen lately for voice have actually been g.shdsl to the premises with an interface converter between that and the pbx. You can always rely on your telco to do what they need to do for the minimum investment possible, and if that means using fiber instead of copper or visa versa then that's what they'll do. As long as the media can support the bandwidth it almost doesn't matter what goes on inbetween, as long as it comes out in the right format (eg T1 in your case) at each end. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
