[asterisk-users] qozap: t3 timer expired for span ...
Hi all message: qozap: t4 timer expired for span 2 qozap: t4 timer expired for span 3 qozap: t3 timer expired for span 2 qozap t3 timer expired for span 3 wow -- what does this mean!? all of a sudden showing up on my server ... no change after reboot .. Junghanns QuadBRI card in place affecting outgoing faxing?! (between bridged TDM400 analog card and QuadBRI) Not a clue why this is .. incoming/outgoing voice calls work, incoming faxes even work but when outgoing fax is dialed, says no one is availale to answer at this time The error has not ever been there before and as far as I know, no isdn wiring has been changed or anything ideas, appreciated! -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content 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
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
thanks for your helpful investigation! I await news :-) -- Chris - Original Message - From: Matt Brown [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 20, 2007 7:55 AM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content 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 -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content 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
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS - Sangoma|Asterisk - POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. I have tried busydetect and busycounts and a number of settings are enabled for UK CallerID support (polarity switch stuff) ... but I had some sketchy side effects with busydetect etc and am wary of premature hangups Thanks for your query -- Chris - Original Message - From: Ed W [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 19, 2007 1:26 PM Subject: Re: [asterisk-users] Disconnect Supervision UK / BT solution? Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact disconnect problem that you see? I have three TDM cards in two different systems, one using PBX lines and one on a private BT line. Both of them have trouble detecting a caller who is ringing, but then hangs up before being answered by the asterisk system However, *all* of them are absolutely fine at spotting a normal hangup once the call is connected and I see no random disconnects during calls either. Can you confirm that this is what you mean, or whether it's something else? Ed W -- This message has been scanned for viruses and dangerous content and is believed to be clean. -- This message has been scanned for viruses and dangerous content 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] Disconnect Supervision UK / BT solution?
Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content 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] asterisk-stat display problems
Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- 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
Re: [Asterisk-Users] asterisk-stat display problems
yep I don't know exactly which things the php-gd is used for, but like I said, someof the pages work, like the main record page, the little red bars showing call volume work fine Really annoying, cause it looks so good at that point, then you go to use the other pages/features and it's broken Thanks for the reply, -- Chris - Original Message - From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 26, 2006 1:49 PM Subject: Re: [Asterisk-Users] asterisk-stat display problems do you have the php-gd package installed on your * server? Chris Earle (CBL) wrote: Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since -- Chris - Original Message - From: Chris Earle (CBL) To: Areski Sent: Tuesday, June 13, 2006 6:15 PM Subject: Re: CDR-Analyser version question Thank you for the reply; I see now that the main file cdr.php does work with argument ?s=1, 2, etc but when s=0, does not load I get an Apache error : relocation error: /usr/lib/php4/20020429/gd.so: undefined symbol: gdFontCacheShutdown Not sure if that means anything important; Also, in the new Asterisk-Stat feature pages like Calls Compare (s=2), the pages do not complete their output -- no search button displayed, stops outputting radio buttons for UserField row etc So at this point, only the main Call-log page (s=1) works. I am using Debian with php 4.4.1 Mysql ver 12.22, Distrib 4.0.24 GD Library is 2.0.33 I think Any input you can pass along would be much appreciated! I am comfortable with php so if you want me to modify sourcecode that is fine Thanks! - Original Message - From: Areski To: Chris Earle (CBL) Sent: Sunday, May 28, 2006 7:11 PM Subject: Re: CDR-Analyser version question No there is no asterisk requirement to make asterisk-stat. Indeed the soft is only based on the cdr database. If you have an error you can give me more info, I may help you. Rgds, Areski On 5/25/06, Chris Earle (CBL) wrote: Hi there, quick question: Does asterisk-stat v2.0.1 require Asterisk 1.2+ ? I am using Asterisk 1.0.x and can't get it to load the cdr.php properly so I downgraded to v1.3 and it works... Let me know if there's an asterisk version requirement for each version of the CDR Analyser Thanks! -- Chris Earle snip -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] ISDN in Japan?
Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with ...INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 so does that mean I can't use a Digium card it there? Can someone please clarify what sort of system I'm looking at here and if I need a japanese retailer for the card or what ;-) Thanks! -- Chris Earle System Solutions Specialist -- 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
Re: [Asterisk-Users] ISDN in Japan?
So it could be BRI, PRI or maybe even Analog there?? I guess what I'm asking is it predominantly ISDN there or not Thanks for the input about the card and chan-capi :-) -- Chris - Original Message - From: Armin Schindler [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 18, 2006 1:32 PM Subject: Re: [Asterisk-Users] ISDN in Japan? On Tue, 18 Apr 2006, Chris Earle (CBL) wrote: Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with ...INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 so does that mean I can't use a Digium card it there? Can someone please clarify what sort of system I'm looking at here and if I need a japanese retailer for the card or what I don't know the status of ISDN in Japan, but the Eicon DIVA Server cards (BRI and PRI) are provided with firmware for ISDN protocols in japan. Together with chan-capi it is fully functional with Asterisk/OpenPBX. Armin -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] Junghanns and Digium TDM400?
Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear results/solutions -- Chris Earle System Solutions Specialist -- 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
Re: [Asterisk-Users] pickup problem
Ha -- this looks useful Just was trying to do a *8 on an IAXy phone...realized it didn't work across protocols If I implement this, I'll have to code in *8 into my extensions.conf instead of relying on the default built in 'steal' ? -- Chris - Original Message - From: Mimmus [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Monday, March 20, 2006 1:17 PM Subject: RE: [Asterisk-Users] pickup problem PickUp2: http://linux.thorsten-knabe.de/asterisk/pickup.jsp works very well. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Monday, March 20, 2006 4:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pickup problem On 20 Mar 2006, at 15:39, Rich Adamson wrote: Mimmus wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, March 20, 2006 4:06 PM there is also a more generic call pickup using 'callgroup=2' and 'pickupgroup=2' in your sip definitions. That approach uses *8 or *8# to pickup any ringing phone within the callgroup number (eg, 2 in this example). Does this call pickup work with IAX2? If yes, how, if there is no callgroup/pickupgroup setting in iax.conf? More in general: does call pickup work between different protocols? Never had a need to do pickup with iax, so don't have a clue. As I recall, the callgroup keyword only applies to sip and zap channels. It doesn't work between protocols. Tim Panton ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] module load order for Junghanns qozap and TDM card
Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- 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
Re: [Asterisk-Users] module load order for Junghanns qozap and TDM card
Maybe this will shed some light about what I'm trying to do: This is some output from dmesg after this load order: modprobe zaptel insmod wcfxs insmod qozap Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) PCI: Enabling device 02:01.0 ( - 0003) qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ 100 CardID 0 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: D-channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: D-channel (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) any thoughts? Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 10:09 AM Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM card Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- 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 -- 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
Re: [Asterisk-Users] module load order for Junghanns qozap and TDMcard
Okay think I finally figured this out it's the modules.conf post-install lines that run ztcfg You're not supposed to run ztcfg more than once with the multiple zaptel cards in there I kept running it manually (ztcfg -) not realizing that after modprobe wcfxs the ztcfg was being run. So the order that works is zaptel qozap wcfxs (which runs ztcfg, and readies asterisk to run) If anyone has any comments about this, please post -- Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 11:30 AM Subject: Re: [Asterisk-Users] module load order for Junghanns qozap and TDMcard Maybe this will shed some light about what I'm trying to do: This is some output from dmesg after this load order: modprobe zaptel insmod wcfxs insmod qozap Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) PCI: Enabling device 02:01.0 ( - 0003) qozap: Junghanns.NET quadBRI card configured at mem 0xf889b000 IRQ 17 HZ 100 CardID 0 qozap: S/T ports: 4 [ TE TE TE TE ] qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: D-channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: D-channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: D-channel (Default) (Slaves: 12) Channel 13: FXO Kewlstart (Default) (Slaves: 13) Channel 14: FXO Kewlstart (Default) (Slaves: 14) Channel 15: FXO Kewlstart (Default) (Slaves: 15) Channel 16: FXO Kewlstart (Default) (Slaves: 16) 16 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) any thoughts? Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 16, 2006 10:09 AM Subject: [Asterisk-Users] module load order for Junghanns qozap and TDM card Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I still need the Digium module. (modprobe wctdm) I've tried a few different orders. Sometimes I can get the digium to load, and the qozap. but then I get an error on the ztcfg about Span invalid argument (could be my zaptel.conf I realize...) *If* I try loading the wctdm after the zaptel and qozap, the server freezes! Some loop about qozap - dropped audio card I don't know if the quadBRI and the TDM are conflicting/sharing the zaptel module, or if I need to modprobe zaptel before each of them? and in what order? Any suggestions appreciated... I haven't even got to figuring out what I can do with chan_capi, just want to get the BRI card on and stuff. Thanks for any ideas! -- Chris Earle -- 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] qozap drops -- possible to bridge BRIstuff ISDN to analog zaptel phone?
Hi all, Using Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q with a Junghanns quadBRI (2 spans connected) and a Digium TDM400 for extensions Shouuld I be worried about these lines that keep showing up in my /var/log/messages? qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 qozap: dropped audio card 1 cardid 0 bytes 6 z1 50 z2 28 qozap: dropped audio card 1 cardid 0 bytes 6 z1 59 z2 37 .. This may or may not be related to a problem I am having with incoming calls. They seem to appear when my incoming calls keep getting dropped as soon as you pick up the analog handset to bridge. Call comes in, rings analog handsetsbut then everything hangs up as soon as one of the dialed handsets is picked up. Stumped on this ISDN dialplan stuff...new to me I've seen mention on voip-info about an edit to the qozap source -- http://www.voip-info.org/wiki/index.php?page=zaptelBRI But that doesn't seem to be my exact problem, as system is loading, asterisk runs..can dial out too! suggestions appreciated! -- Chris Earle System Solutions Specialist ___ --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] Junghanns, Germany ISDN settings
Thanks for the info, I am confused still ;-) It sounds like I need NT mode -- there are NTBA boxes involved at my location... And then -- what do I do about Termination of S/T Interface ?? and Power Feeding? http://www.junghanns.net/downloads/quadbrijumpersnew.pdf That's what I'm referencing Someone feed me some tips please! - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 11, 2006 5:08 AM Subject: Re: [Asterisk-Users] Junghanns, Germany ISDN settings Hi Chris, Chris Earle (CBL) wrote: I've got a Junghanns QuadBRI card which I'm going to install on a system in Germany Anyone give me some tips on the Jumper settings? I'm guessing it's going to be NT mode with p2p? I haven't used ISDN before. I'm going to also put a Digium TDM400P card in there to plug the analog phones into. I'm just worried about the jumpers and modes. It really depends what you will be hooking up to the asterisk box. If you are connecting to a telco's S0 bus you want the card to be in TE mode (Terminal Equipment). If you are using multiple ISDN lines that are coupled together as one bundle (ask the telco) you will probably neet to configure it as p2p. If all lines are singular, use p2mp. If you will be connecting to a PBX, everything is dependant on how that PBX is configured. Best regards, Florian ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] Junghanns, Germany ISDN settings
Hi all, I've got a Junghanns QuadBRI card which I'm going to install on a system in Germany Anyone give me some tips on the Jumper settings? I'm guessing it's going to be NT mode with p2p? I haven't used ISDN before. I'm going to also put a Digium TDM400P card in there to plug the analog phones into. I'm just worried about the jumpers and modes. Suggestions appreciated, -- Chris Earle System Solutions Specialist -- 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] Cannot boot machine up after working on zaptel....
Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over -- Chris Earle System Solutions Specialist -- 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
Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....
Thanks Yeah, you would think so wouldn't you. Tried that , and still wouldn't boot Really annoying. beacuse I've been doing work with the zaptel drivers and such and this happened once already... Thanks for the suggestion, Chris - Original Message - From: Colin Anderson [EMAIL PROTECTED] To: 'Chris Earle (CBL)' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 10:05 AM Subject: RE: [Asterisk-Users] Cannot boot machine up after working on zaptel What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -Original Message- From: Chris Earle (CBL) [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over -- Chris Earle System Solutions Specialist -- 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] Auto login via Remote User
This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER requires web server authentication right? Correct me if I'm wrong, but this means that the contents of the User Table would have to be in the passwd file defined through the .htaccess file right? .. because it passes whatever the user authenticates on the webserver with through to the wiki/extensions scripts right? (resulting in this not really being useful because i've got to duplicate my user table ) Comments appreciated, -- Chris Earle System Solutions Specialist -- 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
Re: [Asterisk-Users] Auto login via Remote User
IGNORE, mispost! :-S - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 4:59 PM Subject: [Asterisk-Users] Auto login via Remote User This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER requires web server authentication right? Correct me if I'm wrong, but this means that the contents of the User Table would have to be in the passwd file defined through the .htaccess file right? .. because it passes whatever the user authenticates on the webserver with through to the wiki/extensions scripts right? (resulting in this not really being useful because i've got to duplicate my user table ) Comments appreciated, -- Chris Earle System Solutions Specialist -- 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 -- 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
Re: [Asterisk-Users] indications issues in Singapore?
Good to know about that Loopstart thing --- helped me quickly solve my problem of the phones not ringing :-) thank you for the input Chris - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 17, 2006 7:10 PM Subject: Re: [Asterisk-Users] indications issues in Singapore? Chris Earle (CBL) wrote: Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I don't run into that problem :-) Analog lines here are mostly loopstart, so you need to enable busydetect if you're using the zaptel FXO. A better option is to use a capi ISDN BRI card. I used the Fritz! PCI card with chan_capi, costs around S$160. The original poster on voip-info wrote about using kewlstart and CPC, which I have never encountered over here. I guess it was in vogue during the good old DID analog trunk days. But nowadays, you either use plain analog or move to BRI/PRI if you need MSN/DDI. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] indications issues in Singapore?
Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I don't run into that problem :-) Any tips appreciated, -- Chris Earle System Solutions Specialist -- 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] Issues in Australia? Ringing, iaxy etc
Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use of notransfer=yes in the iax.conf for the iaxy entry Basically, just hoping for a smooth transition over to the asterisk system Cheers -- Chris Earle System Solutions Specialist -- 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
Re: SV: [Asterisk-Users] BAD/GOOD Echo Cancel
Okay -- I've got serious echo problem with my BT line in the UK, and a TDM400P So I'm starting to consider getting the TDM2400 instead, to take advantage of its hardware echo cancellation -- is this a reasonable idea? I'm all analog, so the TExxx cards aren't an option Comments appreciated, Chris Earle System Solutions Specialist - Original Message - From: Jerry Jones [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 06, 2006 8:19 AM Subject: Re: SV: [Asterisk-Users] BAD/GOOD Echo Cancel On Feb 6, 2006, at 5:04 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Yes. I turned off all echo can on the wildcards and bought external. Point towards carrier and works like a charm. btw - use acoustic echo can not hybrid ec. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
Re: [Asterisk-Users] Fritz card technology German *
Hi all -- I just sat down to revisit this project and figure out what card to go with. Something that no one around me seemed to consider was what Legacy PBX is in place already. There is a PBX with analog phones going into it! So if I get a 4x BRI card, I'm going to either need to keep the Legacy PBX in place and *somehow* connect to the two machines, or I'm going to need a TDM card in the server as well and handle everything inside the one server with Asterisk. This is driving me up the wall, because I'm not physically in the same location as the German PBX/where the Asterisk server is going to be setup. I am also not clear on how ISDN works with incoming and outgoing lines. Does it distinguish between the incoming lines/outgoing? Meaning -- will plugging 2x CAT5 lines between the BRIcard/Asterisk and the Legacy PBX to enable bridging between the two?? Any further comments on this would be appreciated Man is ISDN wacky :-) I'd much rather just get a large TDM2400 and put all the extension lines into that. Cheers all - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 19, 2006 11:59 AM Subject: Re: [Asterisk-Users] Fritz card technology German * Chris Earle (CBL) wrote: I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? Yes. Beronet and Junghanns both have the same cards. (they just 'work' different, junghanns uses zap interfaces, beronet mISDN) So, as already mentioned, you have 2 good options: - 4x BRI card (Beronet or junghanns) - 2x HFC PCI card (uses zap, and are cheap!) Regarding the phones, I only use sip phones, so no idea on that.. ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
Hey all would you believe this problem is still going on? I have now successfully secured individual IRQs for the two TDM cards, but no change has been observed. No interrupts appear to be getting dropped or anything along those lines. Also tried moving the FXO and FXS modules around on the cards. No changes -- not completely clear if the card is defective or not. Can you shed any light on that? I am using Zaptel 1.2.2, with the MG2 echo canceller, but only with Asterisk 1.0.9. I am not prepared to upgrade to a more recent 1.2 version unless I think that will provide massive benefits (I don't think it will affect my zaptel/echo problem?) Back to the card situation: we have not tried any other machines or cards. We are starting to consider using a different card -- and would prefer to stick with Digium hardware. The newer, larger, TDM2400 is under consideration because it has echo cancellation onboard --- Do you think this will do anything?? Also, tried the phone on the phoneline straight up -- sounded fine. We tried the phoneline with a US phone and also a UK one. So is hardware EC the solution to my woes? Chris - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: Chris Bagnall [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 5:24 PM Subject: Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN Thanks for that, insightful The quality on both ends with those settings is quite good -- but the extension side still has a ridiculous -- almost duplicate -- echo! If I turn the txgain right down, I lose all sound.of course the signals in ztmonitor show up perfect then, but can't hear anything (DTMF tones too low or whatever...) Still wondering if it's an impedance issue...or something along those lines/chipset..etc I'm going to attempt upgrading Zaptel now, without upgrading the asterisk Chris - Original Message - From: Chris Bagnall [EMAIL PROTECTED] To: 'Chris Earle (CBL)' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, January 22, 2006 9:13 AM Subject: RE: [Asterisk-Users] Re: Connection TDM400P to UK PSTN Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. Suggestions / Experiences in UK appreciated Most of our clients with BT lines tend to have ISDN BRIs, but we do have one in Northampton running 3 analogue lines from a TDM400. zaptel.conf is as follows: fxsks = 1-3 loadzone= uk defaultzone = uk The TDM driver is loaded with opermode=UK and the output from dmesg confirms this. Relevant settings from zapata.conf are as follows: echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=8.0 txgain=-4.0 busydetect=yes group = 1 context = inbound channel = 1-2 group = 2 context = inbound channel = 3 They're running Asterisk/Zaptel 1.0.10. There were major echo issues when we first deployed the system back in September, but some careful tweaking of rxgain and txgain seems to have largely resolved the situation. Certainly my experience has been that rxgain and txgain have far more impact on echo reduction than any of the echo-specific settings. Get the gains right first, then play with the echo-specific settings. Regards, -- 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
Re: [Asterisk-Users] S100-FX v2.0
Curious about this device as well. Seems almost too good to be true? The built-in switch feature would be much handier than having to get a router for the extra network connection; Also, the 'life line passthru' thing seems interesting -- although I have no idea what a life line passthru is! haha It would be amazing if it was an FXO, but obviously isn't -- I'm assuming you can dial out on that line if necessary? 'life line' concept? Anyone had a good experience with one of these IN THE UK?? (or any country outside of North America for that matter) power issues etc Regards - Original Message - From: Mike Hammett To: asterisk-users@lists.digium.com Sent: Friday, January 27, 2006 1:47 AM Subject: [Asterisk-Users] S100-FX v2.0 I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out and what their opinion of it was. Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- This message has been scanned for viruses and dangerous content 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
Re: [Asterisk-Users] Need to recompile * after changing zap echo method?
Hi there, Don't think you have to recompile * if you've already compiled * with zaptel before. (chan_zap.so exists) Should just have to rebuild zaptel, install the module, and do a ztcfg Good luck, - Original Message - From: Brent Torrenga [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 30, 2006 11:34 AM Subject: [Asterisk-Users] Need to recompile * after changing zap echo method? Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
Re: [Asterisk-Users] Re: Point to Point with Fritz Card ...
I am venturing down this path soon, trying to decide if I will be okay sticking with 2 x Fritz! PCI cards or if I should really get a different card... 4 port ISDN or something... Chan_Capi with 2 Fritz cards I'm told requires code modifications to work etc...? Suggestions? Chris Earle, System Solutions Specialist - Original Message - From: Dias Badekas To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 20, 2005 7:23 AM Subject: [Asterisk-Users] Re: Point to Point with Fritz Card ... I just set up a system with two ISDN pci cards and am using mISDN, plus chan_misdn (multipoint only).It seems to work fine except for a few annoyances, as I wrote in another post.I tried to ran chan_capi, afterwords, just to check on the difference but had problems. Of course, I did not load chan_misdn and chan_capi together as they are mutually exclusive, as per docs.Have you been successful in running chan_capi using the mISDN drivers? The misdn docs say you should be able, but after trying once I wouldlike to hear experiences on this. Chan_capi has a lot of features plus fax stuff implemented that make it interesting.DBOn Tue, 2005-09-20 at 13:15 +0800, Craig Guy wrote: You will need to use the mISDN drivers - the AVM CAPI drivers will not support PTP. It is possible to use mISDN with chan_capi but chan_misdn would be easier. Craig - Original Message - From: "Joao Correia" [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, September 20, 2005 4:57 AM Subject: [Asterisk-Users] Point to Point with Fritz Card ... Hello all, Does anyone has any experience with Point to Point Fritz Card and Asterisk ? I have a BRI access Point to Multipoint working fine but I can only have 3 numbers. The phone telco said that if they change to Point to Point I can have 10 numbers. Does anyone has any experience with Point to Point ? Best regards Joao Correia ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dias Badekas [EMAIL PROTECTED]Athens International Airport -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- This message has been scanned for viruses and dangerous content 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] Newer version of Zaptel with 1.0 branch of *
Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone -- Chris Earle System Solutions Specialist, -- 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
Re: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of *
Has upgrading to the newer Zaptel allowed you to use the newer improvements in it? (sorry if that was implied) Thanks for the speedy reply Chris - Original Message - From: Adam Robins [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 23, 2006 5:01 PM Subject: RE: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and 1.2.1 for the same reasons as you. However, if you ever need to go recompile Asterisk, then you will first need to recompile the old Zaptel, compile Asterisk and the new Zaptel again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Earle (CBL) Sent: Monday, January 23, 2006 4:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Newer version of Zaptel with 1.0 branch of * Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone -- Chris Earle System Solutions Specialist, -- 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
Thanks for that, insightful The quality on both ends with those settings is quite good -- but the extension side still has a ridiculous -- almost duplicate -- echo! If I turn the txgain right down, I lose all sound.of course the signals in ztmonitor show up perfect then, but can't hear anything (DTMF tones too low or whatever...) Still wondering if it's an impedance issue...or something along those lines/chipset..etc I'm going to attempt upgrading Zaptel now, without upgrading the asterisk Chris - Original Message - From: Chris Bagnall [EMAIL PROTECTED] To: 'Chris Earle (CBL)' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, January 22, 2006 9:13 AM Subject: RE: [Asterisk-Users] Re: Connection TDM400P to UK PSTN Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. Suggestions / Experiences in UK appreciated Most of our clients with BT lines tend to have ISDN BRIs, but we do have one in Northampton running 3 analogue lines from a TDM400. zaptel.conf is as follows: fxsks = 1-3 loadzone= uk defaultzone = uk The TDM driver is loaded with opermode=UK and the output from dmesg confirms this. Relevant settings from zapata.conf are as follows: echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=8.0 txgain=-4.0 busydetect=yes group = 1 context = inbound channel = 1-2 group = 2 context = inbound channel = 3 They're running Asterisk/Zaptel 1.0.10. There were major echo issues when we first deployed the system back in September, but some careful tweaking of rxgain and txgain seems to have largely resolved the situation. Certainly my experience has been that rxgain and txgain have far more impact on echo reduction than any of the echo-specific settings. Get the gains right first, then play with the echo-specific settings. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] Connection TDM400P to UK PSTN (Gains and Impedance)
re: setting zaptel to UK mode Yes, I've set the tonezone, and also when loading the module have set it to OPERMODE=UK Looks like that is working A few ideas: *Maybe* the opermode code in the zaptel drivers isn't actually working ? Meaning, maybe it's saying it's in UK mode but actually isn't? Is there any way to know really? Another question: Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? Thanks for your suggestions everyone Chris - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED] Sent: Thursday, January 19, 2006 1:02 PM Subject: Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN On Thu, Jan 19, 2006 at 11:30:25AM -0500, Chris Earle (CBL) wrote: Okay, sorry to hash out this discussion again, but it's starting to drive me crazy Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. I have done tweaking of the gains in North and South America, and after a bit of work have gotten echo to go away, but this seems to just not want to go away. On an incoming call from the POTS, everything on my end sounds perfect, but on the internal extension phone, there is an echo when you speak. An almost perfect copy of what you say. If I turn down the gains on that channel, it doesn't seem to do much, or causes other volume issues. Help! In my research and hunting, I am starting to worry that the US-bought digium cards have IMPEDENCE issues in the UK with the BT Lines etc? That would seem to explain why the echo is so incessant. I have even tried changing Echo Cancellers to MARK3. Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian Suggestions / Experiences in UK appreciated Not sure, but if the card is recent, then its impedence values and such should be configurable. For starters, set the tonoezone (both of them) to uk in zaptel.conf and re-run ztcfg . Try to see what exactly is the model of your card and your firmware. I'm not sure how to check this. Maybe ask Digium support. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is -- 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
Re: [Asterisk-Users] Fritz card technology German *
Thanks for all the posts everyone So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I would rather not have to apply patches just to get the two PCI cards to work in the same box The price difference between the cards you guys mentioned is interesting I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? Thing is, whatever the legacy system in place already is (this is not a fresh operation) must have some sort of minor PBX in place, where all the phones are plugged in. So I would have to remove that and could use a TDM card to plug the phones in? These phones, isdn etc -- probably aren't analog -- probably don't work with a TDM card right? So I think what you were suggesting John is ISDN channel cards and a TDM in the same machine? with * just bridging calls between the two? Interesting. :-S Chris Earle - Original Message - From: John Daragon [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 18, 2006 6:45 AM Subject: Re: [Asterisk-Users] Fritz card technology German * Chris Earle (CBL) wrote: Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card The AVM Fritz card is a single connection (2 x 64 kbps) passive ISDN card. It's well supported by chan_capi, but running more than one of them in a PC requires a driver patch. You can't use a Digium card because Digium doesn't make an ISDN2 card. We have 2 ISDN lines ( -- 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then there's some sort of channel bank that sends the calls out to the extensions. Does this make any sort of sense? By 2 lines I guess you mean 4 channels ? i.e. 4 simultaneous calls ? If you mean 2 channels, then you only need 1 fritz card. Could someone confirm with me that this is the right direction to go -- ISDN lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension handsets.. On the handset side you could use a couple of TDM4xx cards, or just use SIP phones. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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] Re: Connection TDM400P to UK PSTN
Okay, sorry to hash out this discussion again, but it's starting to drive me crazy Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. I have done tweaking of the gains in North and South America, and after a bit of work have gotten echo to go away, but this seems to just not want to go away. On an incoming call from the POTS, everything on my end sounds perfect, but on the internal extension phone, there is an echo when you speak. An almost perfect copy of what you say. If I turn down the gains on that channel, it doesn't seem to do much, or causes other volume issues. Help! In my research and hunting, I am starting to worry that the US-bought digium cards have IMPEDENCE issues in the UK with the BT Lines etc? That would seem to explain why the echo is so incessant. I have even tried changing Echo Cancellers to MARK3. Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian Suggestions / Experiences in UK appreciated -- Chris Earle System Solutions Specialist, - Original Message - From: John Novack [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Wednesday, August 24, 2005 10:03 AM Subject: Re: Connection TDM400P to UK PSTN The jacks on the TDM are ( incorrectly ) referred to as RJ45, correctly they are 8 position modular. The line, either in or out is on the two CENTER pins. NONE of the other 6 pins are used. Though I am not in the UK, from what I know you don't use the two center pins for a single line connection, so you will need to fashion some sort of adapter to connect. Frankly, using the two center pins ( A Bell System brain blizzard) wasn't the smartest idea. It makes the modular plug into, with the addition of just a little moisture, a really good spark gap when a ring signal or small induction of lightning is applied. I have seen many a modular plug turned black and useless since the introduction of modular in the US in the early 70's Good luck John Novack Graham Kiff wrote: I'm a complete Asterisk novice and have an installation based on the [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]CD. I've installed my TDM400P with 2 x FXO 2 x FXS, but every time I try to dial out, I get a message No circuits available. Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to know the pin mappings from a standard BT plug to RJ11. Cheers Graham ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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] Fritz card technology German *
Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card We have 2 ISDN lines ( -- 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then there's some sort of channel bank that sends the calls out to the extensions. Does this make any sort of sense? Could someone confirm with me that this is the right direction to go -- ISDN lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension handsets.. Thanks -- Chris Earle System Solutions Specialist, -- 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] Config files under CVS versioning system
Hi all, I'm about to start building a bunch of asterisk servers with a team of developers and I thought it would be a good idea to put each server's config files under CVS so that we can keep track of changes, revert back etc... Which files do you think I should include in the cvs modules? just the .conf files? AND would it be possible to have the actual server copies be the versioned copesi so all I'd have to do is a cvs update to install new changes? Thoughs and suggestions appreciated, Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?
so, do you think that this upgrade is worth doing, or should I hold out a bit longer (since I am not having any troubles right now) until the next release? I am assuming that there will probably still be a few more upgrade releases... C h r i sE a r l e System Solutions Specialist - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 17, 2003 9:06 PM Subject: [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes? I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved Temporarily message incorrectly after receiving a NOTIFY message. CSCea93969 The Cisco ATA loses G.723 audio when call waiting occurs. CSCeb01064 The Cisco ATA From header domain value changes SRV record name. CSCeb17953 The Cisco ATA stops the registration process if it receives an unexpected response to a REGISTER request. CSCeb19228 The callback-on-busy feature does not work for calls to a PSTN. CSCeb23060 Upon receiving a 4xx response to a REGISTER request from a backup proxy, the Cisco ATA needs to continue retrying the request with the primary proxy . CSCeb24556 The Cisco ATA may fail to send a ring tone when acting as a transfer target in a blind transfer. CSCeb28218 The Cisco ATA, while in a call, detects audio from an incoming call. CSCeb32210 When the SDP attribute a=fmtp appears before the attribute a=rtpmap , the Cisco ATA will not send out-of-band DTMF digits. CSCeb35955 Attended call transfers occur even when this feature is disabled via the PaidFeature configuration parameter. CSCeb36752 Call forwarding does not work when the Cisco ATA detects a busy signal. CSCeb37037 The Cisco ATA stops registering after a 2.16 upgrade is performed. CSCeb37043 The call-waiting default user setting cannot be controlled by the CallFeatures configuration parameter when the Cisco ATA obtains its configuration file from the TFTP server. CSCeb40099 The Cisco ATA plays an incorrect tone after unconditional call forwarding is enabled or disabled. CSCeb44406 Change the behavior of the Cisco ATA to not remove all registrations. Full information can be found here: http://www.cisco.com/en/US/products/hw/gatecont/ps514/prod_release_note09186a00801a2519.html JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: list format vs newsgroup format
Agh I hate trying to sift through all these messages and keep track of the various threads going on . Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... only problem being that this list requires list membership before postingShrug C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: list format vs newsgroup format
what? you want to change the list into a message board? no no, newsgroups are fine... phpbb is nice though... C h r i sE a r l e System Solutions Specialist - Original Message - From: jltaylor [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 18, 2003 10:35 AM Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format IF there was a consideration for a change, I prefer: phpbb it's open source and easy to use. www.phpbb.com you can still get emails from the posts. -- Original Message -- From: Chris Earle (CBL) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 18 Jul 2003 10:02:22 -0400 Agh I hate trying to sift through all these messages and keep track of the various threads going on . Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... only problem being that this list requires list membership before postingShrug C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: list format vs newsgroup format
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 18, 2003 10:16 AM Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote: Agh I hate trying to sift through all these messages and keep track of the various threads going on . Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... What you need is to get a decent mail reader. Those of us that complain regularly about people changing subjects in the middle of a thread already know the benefits of threaded email reading. Why bother with a newsgroup because you choose to stay on windows and use outbreak express hmmI think I *do* realize the benefits of threaded email reading -- that's why I was supporting services like gmane! ;-) I do realize that OE isn't really the best idea, but it works fairly well with some proper setup (folders, rules etc). only problem being that this list requires list membership before postingShrug And this is a good thing. Otherwise spammers only need the list address to spam us all, and you get this also on newsgroups. Right now the only risk is the fact that the email addresses we use here are archived publicly in an easy to harvest method. I think that is the most risk I wish to undertake. Yes yes, I know it's a good thing, just didn't help my newsgroup format/gmane solution...hehe No complaints here about taking measures to prevent spam, it's more than necessary IMO. C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: list format vs newsgroup format
OE has the 'group messages by conversation' option, but I think I wrote it off awhile ago because it didn't work so well or something. maybe I'll give it a second chance C h r i sE a r l e System Solutions Specialist - Original Message - From: James Taylor [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 18, 2003 10:50 AM Subject: RE: [Asterisk-Users] OT: list format vs newsgroup format So, other than Outlook for Win and Xfmail for Linux any recommendations for those lost souls on the list? -- Original Message -- From: John Laur [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 18 Jul 2003 09:17:57 -0500 Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... If you want threads, get a MUA that is capable of threading. Most are. The In-Reply-To header makes mail threading on lists trivial (and you can easily spot the people that hit reply and change the subject without actually starting a new thread...) Maybe you just have not yet found where to turn it on. only problem being that this list requires list membership before postingShrug Which is why the mailing list will probably continue to be used... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Cygwin?
Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
