[asterisk-users] Very high translation costs for g729
Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI show translation Server 1: g729 -26252525252426 -5336 Server 2: g729 -66656565656469 -9075 On my other three boxes, I get much saner vaules (costs anywhere from 3 to 6). Any ideas why two boxes have such high costs? All the servers run the same OS, updated to the same versions of everything, including kernel. Four of the five boxes run x86_64 kernels, with the two that are playing up both running x86_64 kernels. I've switched the entire network to using Speex instead of g729 until I find out why I'm getting such high numbers here. I suspect (but can't prove) that this may have been the cause of some audio issues between these two servers as the phones on either end use alaw, so Asterisk is transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Very high translation costs for g729
On 06/11/2006, at 8:53 AM, Julian J. M. wrote: Try forcing asterisk recalculate those costs: Ok, that fixed it. Thanks! :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Asterisk and ISDN and Hylafax
On 27/10/2006, at 7:22 AM, Thomas Winter wrote: I have to set up an Asterisk with an 4-port BRI card. Hylafax should send and receive fax. Will this work reliable? I have a Eicon V-4BRI (which is in fact a voice-only board) that does faxing via HylaFax/IAXmodem and its flawless. However, its really low- volume (maybe 1 or 2 faxes per day in or out) . I would follow Armin's recommendation and go with the full 4BRI board that has on- board fax capabilities. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] checking 'voicemail externally - doesn't work
On 23/10/2006, at 10:13 AM, Joseph wrote: I'm trying to log-in externally (from PSTN line) to check my voice-mail so I created context to authenticate log-in Just create an inbound route to VoiceMailMain(). Then, press * during the outbound message and it'll prompt you for a password. Hey presto, you're inside your voicemail! cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:24 PM, Martin Joseph wrote: It doesn't work. pressing * during my outgoing message does nothing. Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:26 PM, Eric ManxPower Wieling wrote: The previous poster is obviously running some Asterisk GUI. Yes, sorry. I am running FreePBX, but I didn't notice the | in the call to VoiceMailMain, otherwise I would've mentioned it. :( My bad. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote: Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. FreePBX is NOT Asterisk. Yes, I know that. Hence the 1.2.12.1 *with* FreePBX statement. I.E. Asterisk v1.2.12.1 *with* FreePBX *added* I know what FreePBX is. I also know the differences between Asterisk, FreePBX, [EMAIL PROTECTED] and TrixBox. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Reception Console
On 16/10/2006, at 2:32 PM, Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. If it can handle multiple Asterisk servers -- ME, ME! PICK ME! PICK ME! :) Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] ftp server
On 10/10/2006, at 2:10 AM, Noah Miller wrote: Quite right. I'm blaming the inadequacies of my OS on vsftpd. vsftpd just uses your OS user accounts. On the Tao linux box that I had it installed on, you couldn't do capitals in user account names. My bad. Which is weird, because I thought Tao was like CentOS: A basic rebrand of RHEL. And I use CentOS. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] ftp server
On 08/10/2006, at 3:00 PM, Dean Collins wrote: Whats the best ftp server to upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using BTF server. I'm using vsftpd quite successfully on several Asterisk boxes with Polycom IP501 phones. Though, I'm now considering switching to HTTP provisioning so that I can actually dynamically create Polycom configurations from a MySQL database. At the moment, its all vaporware, but its a nice idea. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] ftp server
On 09/10/2006, at 5:07 AM, Noah Miller wrote: username and password is PlcmSpIp. vsftpd cannot handle capitalized usernames, so if you want to use vsftpd, you have to manually re-configure the username on each phone. I use vsftpd and I'm using the default PlcmSpIp username just fine. :) Essentially, I configured PlcmSpIp as a Linux user and I'm serving it out by using personalised FTP home directories in vsftp and then chrooting per user. Works like a charm and no phone configuration is required. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] Optus PRI via DSL
On 08/10/2006, at 9:34 PM, Paul Hales wrote: I have seen an Optus SHDSL box set up incorrectly before - and the tech re-visited and set it up correctly within hours of being informed. Same with my Optus SHDSL box: The first tech misconfigured, so I kept getting PRI restarts on my Sangoma card. They came back and reconfigured and now it works like a charm. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
RE: [asterisk-users] ftp server
On Mon, October 9, 2006 11:46 am, Dean Collins said: Are you able to track real time from a windows machine the transactions occurring on your asterisk server if you have vsftpf installed? Yes... In an SSH session, tail -f /var/log/vsftpd.log will show you everything you need. Also, I have all my mac address.cfg files in one directory, and then each phone gets its own directory to store its own phone.cfg and log files. Works like a charm. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] polycom reboot script
On 09/10/2006, at 12:12 PM, Dean Collins wrote: can anyone give me an idea on how this reboot script works? I actually just use the SIP notify command on the Asterisk console to remotely reboot my Polycom phones. It requires a pre-configured sip_notify.conf file and the Polycom option to reboot on config check. You can then call it from a script using: # asterisk -rx sip notify polycom-reboot 400 (Where 400 is the SIP ID of the phone). I'm interstate at the moment, but if you send me an email, I can lookup the settings when I'm back on Wednesday. Ta, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] asterisk-addons-1.2.4 Installation Problem
On 05/10/2006, at 4:25 PM, Abdul wrote: But i am little confiuse why i am not able to install MySQL Real- Time. here is the Error when i am trying to make all for asterisk- addons-1.2.4. You need to install the mysql-devel package to get the header files. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [asterisk-users] asterisk to asterisk DID extentions
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the
DID caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
exten = _8XX,1,Dial(IAX2/BoxA/${EXTEN})
Note that I wrote this from memory so its probably not syntactically
correct. I hope you get the gist though. :)
cYa,
Avi
--
National Manager - Special Projects
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Fitzroy, VIC F: +61 (0) 3 9235 5444
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Re: [asterisk-users] Trixbox Documentation
joea, j4computers wrote: So, now I am struggling with a Suse SLES 9 install, that seems reluctant to co-operate. I have a number of boxes running CentOS 4.4 with Asterisk 1.2 and FreePBX: Because I install everything manually, I know it all works, without the overhead of the Trixbox features I have no intention of ever using (a2billing, Sugar, etc). I find that following the FreePBX install procedures for CentOS to be quite straight-forward. I have a bunch of Digium and Sangoma cards as well, all working too. My 2c, YMMV, etc. Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Uninstalling Trixbox
Mike Dent wrote: I'm curious how you installed Trixbox? There is a tar.gz version of Trixbox that can be installed over an existing RHEL4 or CentOS installation. However, removing Trixbox is very difficult. You are better off reinstalling RHEL4 and then installating Asterisk from scratch. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] Noob question: Packet size
Hi guys, I have what is probably a very noob question. I've tried to search the wiki, but my lack of knowledge is hindering me in finding the right keywords: I'd like to know what the packet size of an IAX2 packet is, if its using the ilbc codec. Now I'll tell you why, so you can tell me what I really want to know. :) I'm experiencing packet loss on my inter office network, so I installed SmokePing to determine the extent of the loss. However, I'm not sure what the best size packet to test would be. Any advice/suggestions would be great. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] chan_zap.so stopped working after upgrading CentOS
Brent Franks wrote: We ran into the same thing, and the only way I can get it to work (which is goofy, but it does work) is modprobing the same device multiple times. Try waiting after modprobe zaptel for udev to create the device nodes. I do this: modprobe zaptel wait 5 modprobe wctdm ztcfg -vvv And it works fine for me on CentOS 4.3 cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] latest CentOS-asterisk-freepbx installation procedure
Roland wrote: I've tried all those at voip.info.org but I just couldn't get it right. and I don't have the luxury of time to try figure out how to make it work by myself. The official FreePBX install docs (which have Asterisk instructions as well) for CentOS are here: http://aussievoip.com/wiki/index.php?page=freePBX-Centos cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Kevin P. Fleming wrote: if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. Is there a Wiki page or similar describing how to checkout SVN for Asterisk? Also, will I need to checkout and compile SVN versions of Zaptel/Libpri/Addons (as I use all three)? Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Avi Miller wrote: Is there a Wiki page or similar describing how to checkout SVN for Asterisk? Also, will I need to checkout and compile SVN versions of Zaptel/Libpri/Addons (as I use all three)? Replying to myself to say that I've found Digium's instructions and I'm testing SVN on my test server now. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Avi Miller wrote: Replying to myself to say that I've found Digium's instructions and I'm testing SVN on my test server now. :) And again to say that it seems work just fine with the SVN code. Thanks Kevin! -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] Polycoms, Attended Transfer and Canreinvite = yes
Hey guys, I've been trying to change my Asterisk setups to use canreinvite=yes. I'm having a small problem with my Polycom IP501 phones and transferring calls. If a call comes in via my ISDN BRI lines (using chan-capi), I can successfully transfer the call using the Polycom Blind Transfer option (Transfer - Blind - EXT - Send). However, if I try to use the attended transfer method, the call is never connected to the new user. When I hit transfer, the caller gets MOH and I dial the destination ext. Once the person answers, I hit Transfer Now .. the MOH stops for the caller, but both phones are dead. The call is never reconnected successfully. On the console, I see this: -- Called 405 -- SIP/405-0849cba0 is ringing -- SIP/405-0849cba0 answered SIP/401-084a0ba8 -- Attempting native bridge of SIP/401-084a0ba8 and SIP/405-0849cba0 -- Stopped music on hold on CAPI/V4BRI-2/92355400-25 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/401-084a0ba8ZOMBIE' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/401-084a0ba8ZOMBIE' -- Incoming call: Got SIP response 500 Internal Server Error back from 192.168.1.128 == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'CAPI/V4BRI-2/92355400-25' 405 is the extension I'm trying to transfer the call to. Any advice? I've been searching the list archives and the wiki, but can't find anything specific. Ta, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] Tracing audio problems
Hey guys, I need some assistance in tracking down the cause of audio problems that are occurring at two of my sites: Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both sites are reporting that audio in calls is dropping out during words, so that the other caller (i.e. the remote user) can only hear bits of the words. This used to only happen on Asterisk-to-Asterisk calls via IAX2 (using g729) so I assumed it was latency or bandwidth problems on the inter-office network. However, the network is hardly used and my round-trip times are sub 100ms according to iax2 show peers (with qualify=yes). Then, thinking it might be g729 issues, I changed the entire system to only use alaw and the problem persists. Does anyone have any suggestions on where to look next? My users are getting increasingly annoyed and I'm quickly running out of ideas. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Tracing audio problems
Avi Miller wrote: Does anyone have any suggestions on where to look next? My users are getting increasingly annoyed and I'm quickly running out of ideas. Replying to myself to note that this is now happening on outbound calls via ISDN, i.e. calls that don't use IAX2 or the inter-office network. It also happens on inbound calls. Ta, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Tracing audio problems
On Mon, August 28, 2006 5:17 pm, Erik said: Through what means are both sides connected, 1:1 xDSL? All offices are connected via 512/512 SDSL. What bandwidth, are you using tunnels (pptp/gre/ipsec), how many concurrent calls etc. No tunnels (that I'm aware of). Very few concurrent calls, probably max 2 per location to the head office. You could try analysing network delay/jitter/packetloss using Smokeping. Note that on DSL 1 g729 calls uses about 45 kbit/s, alaw uses about 108 kbit on DSL I'll try that tomorrow, thanks! :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Tracing audio problems
On Mon, August 28, 2006 5:21 pm, Matt Riddell (IT) said: Are you using realtime? No, the Asterisk boxes are managed by FreePBX which creates .conf files. I have two boxes playing up (the ones with PRI connections). My other three servers that use BRI are just fine. Calls between the other three boxes are fine, too. Calls make from one of the BRI-based servers to one of the PRI-based boxes can suffer. Which makes me think -- could this be a Zaptel timing issue? The two PRI boxes each have a Sangoma A101u PRI card and one of them also has a TDM400P with 4x FXO modules. The BRI boxes only have Eicon Diva Server 4-BRI cards, so they use ztdummy. Is there a Zaptel tuning application for the PRI cards? :) Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Tracing audio problems
On Mon, August 28, 2006 8:10 pm, Rich Adamson said: Is this a new installation, or, were the boxes working okay for a while and they just now started having problems? Its not a new installation: Calls have been fine for at least a month on one server and about 4 months on another. Both servers are in our head offices (two separate offices seperated by a 100mbit wifi link, each with their own Asterisk box and ISDN PRI). parameter wasn't right. Check /etc/zaptel.conf for: span=1,1,0,esf,b8zs I'm in Australia with EuroISDN, so I have this: span=1,1,0,ccs,hdb3,crc4 Are the poor audio calls always associated with one site (head office)? One location (our Sydney offices), two sites, two different servers both experiencing the same problem. I suspected an ISDN issue, as they use the same ISDN provider, but my provider assures me (yeah, I know) that there are no faults on either line. What does 'zap show status' indicate at those sites that have bad audio? Server 1: getafix*CLI zap show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 Wildcard TDM400P REV I Board 1 OK 0 0 0 Server 2: rincewind*CLI zap show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 These are with no calls in progress though, so I'll try it again during the day tomorrow to see if anything changes. Do you have iax links to these sites as well, and if so, are you having the same audio problem with them? Yes I do, and yes I am. I have IAX2 links from each server to each other server (5 in total), so there are four IAX2 trunks configured on each box. They used to use g729, but I switched them all to alaw to see if transcoding was causing the issues. I'm told it persists, but I'll know more tomorrow as I expect I'll be on the phone a lot to the other offices. What type of phones are you using to initiate the calls with bad audio (sip phones or what)? All our phones nationally are Polycom IP501 phones. The branches now use the SIP1.6 code, but I suspect the head office phones may still be on 1.5. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycom IP430 won't finish boot
DM wrote: Why do you think the problem may be with the FTP server? I've been running vsftpd on several different systems, all with Polycom's. There were reports that the Polycoms preferred some FTP servers over others, but I also use vsftpd (using the default PlcmsSpIp username/password combo) quite successfully on my five provisioning servers. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] Call to a queue killing Asterisk?
Hey guys, Last week I changed my queues from using proper agents and AgentCallbackLogin() to using the the FreePBX default with fixed agents (which uses the Local/[EMAIL PROTECTED] style for the member= field). I've also upgraded to Asterisk 1.2.10 and FreePBX 2.2.0 Beta 1. Since then, I noticed that my FOP would sometimes get stuck when a call hit the queue (showing all the agents being busy forever, until a op_server.pl reload). I started to track it this morning and actually saw Asterisk shutdown as the call got answered (and get restarted by safe_asterisk, of course). This accounts for the stuck FOP, but now I have the joy of working out why Asterisk is shutting down. I don't see anything in /var/log/asterisk/full -- I see the mysql CDR being recorded and then 4 seconds later, I see the Asterisk startup sequence happening. Anyone have any suggestions on where to start debugging this? Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)
On Fri, August 11, 2006 4:26 pm, Wolfgang Paul Rauchholz said: allow=g729 allow=g723 Do you have the g729 and g723 codecs installed? They are not installed with Asterisk by default. cYa, Avi ___ --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] Correct syntax for Set(CALLERID(all)...
hugolivude wrote: I'm able to get the number to change but the name is always Unknown Name. I've tried numerous combinations of quotes, but just cannot get the name... I use Caller Name401 Note, no space between the closing and the character. Seems to work for me and Polycom phones. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycom 1.6.7 firmware?
Dean Collins wrote: Yep, but didn’t [EMAIL PROTECTED] have a folder to store these files on? Does freepbx? You mean TrixBox? I know they're working on a phone provisioning system, but I thought it was just for Cisco and Grandstreams. Check with the TrixBox guys at http://www.trixbox.org (FreePBX is just a GUI configuration utility. TrixBox is the successor to [EMAIL PROTECTED], i.e. the all-in-one Asterisk-in-a-Can distribution. TrixBox uses FreePBX as part of its management tools). -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycom 1.6.7 firmware?
Stephen Murphy wrote: Can you simply replace your current sip.Id and sip.ver files with the latest firware files or is this dangerous? That's what I did, after doing a diff of the old and new original sip.cfg and phone1.cfg files to make sure there weren't any major changes/additions. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Polycom 1.6.7 firmware?
Stephen Murphy wrote: And this worked without issues? It did for me. YMMV, depending on the changes you made to your sip.cfg and phone1.cfg. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] Fax tone detected, but no fax extension for CAPI
Stefan-Michael. Guenther (in-put GbR) wrote:
I have a fax server with an AVM Fritzcard that is connected to port number 4
of an EICON DIVA Server 4 BRI.
If the inbound is always going to be fax, set faxdetect=off in
capi.conf, so that it just runs the default.
Otherwise, add a fax extension:
[faxout]
exten = _X.,1,Answer
exten = _X.,2,DIAL(CAPI/g1/${EXTEN},10,r)
exten = _X.,3,Congestion
exten = fax,1,Dial(CAPI/g1/${EXTEN},10,r)
That'll dial out the call when it detects its a fax.
hope that helps,
Avi
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Re: [asterisk-users] Setting CALLERID on a residential telco line
Andrea Spadaccini wrote: Is there any hope to change the caller-id on a BRI line? I can change my Caller ID on my BRI lines to anything within my DID range. Hope that helps, Avi ___ --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] Prevent a Polycom contact list to be overwritten
Douglas Garstang wrote: The phone will quietly not be able to write to the contacts directory. However, it seems the directory on the phone is maintained. I still can't work out how to get the Polycoms to replace any locally added directory items with a master list from the provisioning server. cYa, Avi ___ --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] Polycom config file location
Stephen Murphy wrote: My question is: How do I get the current config files the phone is using off the phone? AFAIK, you can't. :( You can only provide new configuration files from your FTP/TFTP server. However, the Polycoms do strange things when they've been configured in multiple locations. You might find the phone overwriting the configuration files with its original configuration. That is not confirmed though. I've just seen my Polycoms do weird stuff in the wild. :) -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9235 5400 3065 F: 03 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] sending out fax using asterisk
root linux wrote: I am having problem sending out fax from fax using an ATA connected to the asterisk. Your system is detecting the fax and trying to receive it at the same time. I had the same problem for a while, and Armin nicely changed chan_capi for me. :) Essentially, if you're using Zaptel, change zapata.conf to have faxdetect=incoming instead. That way, it'll only do fax detection on incoming calls and not on this outgoing call. The same line now also works in chan_capi for calls made by that channel driver. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] sending out fax using asterisk
root linux wrote: I am not using any Zaptel card... I am doing a back-to-back to Verso C5CM via Internet Wow. You're going to probably run into problems trying to fax over a VoIP connection. Other people can explain why far better than I can. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [asterisk-users] 2 NICs; Asterisk receives on eth1 and replieson eth0
Douglas Garstang wrote: Yes, we tried to do the same thing. We wanted our Asterisk system to be multi-homed. My head office Asterisk box is multi-homed: I have three networks across two NICs. One dedicated to hardphones, another to the local LAN (and PC-based softphones). The third network is bound to the same NIC as the LAN, but has different IP addressing. This links to our national VPN to connect to Asterisk boxes in other cities. All of the regional Asterisk boxes are also multi-homed. They have two IP addresses (sometimes on one NIC, sometimes on two). One connected to the local LAN, the other to the national VPN. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] ISDN (E1) Hardware Echo Cancellation
Hey guys, Could someone recommend some good hardware echo cancellation devices for a single ISDN E1 line? I need something to sit between the wall and a Sangoma A101u PCI card. Preferably a device that I can source in Australia! :) Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] ISDN (E1) Hardware Echo Cancellation
Doug Lytle wrote: A Tellabs 2572 64ms EC. Check ebaY. Instructions on the Wiki. Anything that requires a little less soldering? :) I was hoping for a boxed solution. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Polycom Intercom - almost there
Bill Gibbs wrote: Any ideas or suggestions? Just trying to keep the number of button presses to a minimum. The number of button presses would be the same though: 1. Pick up the phone, dial 7 3 0 0 (four buttons) 2. Hit line 3, dial 3 0 0 (four buttons) You could configure the line 3 button as a Speed Dial to a prompt that asks for an extension. Then, it pages that extension, perhaps. If you use background(), it could be as quick as pressing the button and immediately dialling the extension. Just some ideas. :) -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9486 0411 3065 F: 03 9486 0611 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Debian Sarge or CentOS4.3
Tom wrote: I don't like the fact that CentOS is nothing more than a copy of RH Enterprise Server. It is almost like running a Windows clone. I would rather find and run something better. While I love CentOS for the very same reason: I get all the benefits of Red Hat Enterprise Linux without the annual fee. :) All of my FreePBX boxes run on CentOS. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9235 5400 3065 F: 03 9235 5444 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Voicemail with NFS
Douglas Garstang wrote: I don't think unison is a workable solution. It doesn't scale. The network and system load would increase exponentially as we added asterisk servers to our cluster. If you're clustering that many boxes, I'd investigate fibre channel SAN and GFS. That way, each node of the cluster just mounts the voicemail location locally. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9486 0411 3065 F: 03 9486 0611 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Digital Receptionist
Khaled Chehab wrote: Hi I make a Digital Receptionist ,but how can I attach it to an extension [EMAIL PROTECTED] is now called TrixBox. You'll get a lot more support at http://www.trixbox.org cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9486 0411 3065 F: 03 9486 0611 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] what are the elements of a good asterisk setup?
Tyler Retzlaff wrote: to use Active card(s) and I need to accommodate 2 x BRI TE/PTP services. I'm devoted to Eicon Diva 4-BRI cards: They're expensive, but they have onboard DSPs and Echo Cancellation, which is awesome. Also, great Linux driver support and chan_capi-cm support for Asterisk. The source driver set from Melware will happily compile on Fedora Core (I use CentOS myself). cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9486 0411 3065 F: 03 9486 0611 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] [EMAIL PROTECTED] / Trixbox Question
Johnny Stork wrote: CentOS behind trixbox is a relatively complete CentOS system? The installation of CentOS is sufficient to support TrixBox, but you can always add additional packages using yum. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] transfer other features
Ronald Wiplinger wrote: What do I miss ??? Your current blind transfer setting is ##, so try ## 632 instead. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] Configuring Polycom 501 IP phones via the console
Stephen Bosch wrote: All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. The console is very tedious. Why not use the web interface instead? Let the phone get an IP address via DHCP and then point a web browser at the phone. :) Much easier to navigate/configure. Password is the same as the advanced password on the phone itself. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Web based interface
Kerry Garrison wrote: There are several listed at http://voip-info.org. For Management check out FreePBX, for recorded calls look for Asterisk Recording Interface. FreePBX includes ARI, btw. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Polycom 301's drop last two digits of dialed number
On 26/05/2006, at 7:49 PM, Jamie Heckford wrote: Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Polycoms have their own dialplan built into the phone. Depending on how you configure your phone (i.e. on the phone, or via the web interface or via FTP), you will have modify the onboard dialplan to allow numbers longer than 10 digits. Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] Voice Mail Audio Progression
On 25/05/2006, at 8:14 PM, Bob Chiodini wrote: message that says Asterisk mail then short pause then the word mailbox then a very long pause, then a request for a password. I Its asking you for your mailbox number at that point, then pausing to allow you to enter the mailbox number. When you don't, it assumes you mean the mailbox associated with the extension you're dialling in from. Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] Voice Mail Audio Progression
On 25/05/2006, at 8:57 PM, Bob Chiodini wrote: I don't hear a request for my mailbox number. Should it say something like Enter mailbox number? I believe the prompt just goes Mailbox? -- its not great. But, there's no other prompts being played in your output. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] macro-dial
Mimmus wrote: I'd like to drop this script: does anyone can explain me what is its main job? Dialparties.agi is used to test all of the submitted destinations for Call-Waiting and Call-Forward settings before passing the final extension(s) that can be called back to Asterisk. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] latest @Home questions
Michael George wrote: We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. You're probably better off asking at the FreePBX forums (http://forums.freepbx.org). In answer to your question: The default behaviour for MeetMe changed for FreePBX: It no longer creates a conference for every extension. Rather, you have to manually create all the conferences you want on your system (using the Conferences option, as you've discovered). I suspect you're working off old [EMAIL PROTECTED]/AMP documentation. The new user documentation for FreePBX can be found at http://www.aussievoip.com.au/wiki/index.php?page=freePBX Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] how to customize voicemail
On 22/05/2006, at 9:13 PM, [EMAIL PROTECTED] wrote: Is it a way to record a welcome message and use it ? Dial into VoiceMailMain() and hit 0 for Mailbox options. You can record both an Unavailable and a Busy message. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] Quad BRI card
On 18/05/2006, at 6:51 PM, Wayne Gemmell wrote: are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! But worth every penny, imo. I have a few servers running Eicon Diva Server V-4BRI cards and they are easy to install, run great with Armin's chan_capi-cm and the onboard hardware echo cancellation is excellent. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.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
Re: [Asterisk-Users] no SUBSCRIBE request sent
On 17/05/2006, at 7:36 PM, richard Coco wrote: [local] exten = 2001,1,Dial(SIP/2001,10,tr) exten = 2002,1,Dial(SIP/2002,10,tr) [notify] exten = 2001,hint,SIP/2001 exten = 2002,hint,SIP/2002 Try this: [local] exten = 2001,1,Dial(SIP/2001,10,tr) exten = 2001,hint,SIP/2001 exten = 2002,1,Dial(SIP/2002,10,tr) exten = 2002,hint,SIP/2002 cYa, Avi ___ --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] no SUBSCRIBE request sent
On 17/05/2006, at 8:27 PM, richard Coco wrote: unfortunately i still don't see subscribe request in the sip debug trace. Have you configured your phone to subscribe to the extension? :) cYa, Avi ___ --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] mISDN FAX
On 17/05/2006, at 1:26 PM, MBIT Technologies wrote: I have mISDN installed and working correctly but I am unable to receive a fax through the connection. I have NVFaxDetect and RxFAX running on my CAPI channels, so I know that works. :) Does NVFaxDetect detect the fax correct? Does it drop to the fax = exten? cYa, Avi ___ --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] mISDN FAX
On 17/05/2006, at 1:45 PM, MBIT Technologies wrote: I can't see any fax detection at all in my call logs. What does your dialplan look like for incoming calls? Do you give NVFaxDetect enough time? I find that 4 seconds is good, but 2 seconds is dodgy, for example. cYa, Avi ___ --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] mISDN FAX
On 17/05/2006, at 2:29 PM, MBIT Technologies wrote: This is what Im getting when I try to receive a fax Yeah, looks like NVFaxDetect isn't dropping to the fax extension. You may want to check with the NV guys to see if it works with mISDN. For reference, I have it working with the Eicon Diva Server V-4BRI (using DIVAS4LINUX + chan_capi-cm) and the Sangoma A101u (using WANPIPE and Zaptel). cYa, Avi ___ --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] CallerID retain on internal transfer
Michael J. Tubby B.Sc (Hons) G8TIC wrote: call then transfers it on to another extension transferee (recipeient) sees the Caller*ID This behaviour changed in Asterisk 1.2 -- add o to your Dial options and Asterisk will retain the original Caller ID on transfer. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Eicon Diva Server - Fax and data modem support
On 11/05/2006, at 6:03 PM, Isaac Xiao wrote: Would any one advice how implement Diva Server BRI or PRI card to support fax and data modem? In Eicon’s website, it says that they support these. But there is no FXS port on the card, how it can be connected to Fax machine or data Modem? It *is* a Fax Machine and a Data modem. :) cYa, Avi___ --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] FreePBX in production?
Time Bandit wrote: And the *_additional.conf files are the ones overwritten by the config in the DB. So you can edit the other ones. You could, but it'll get overwritten by any FreePBX upgrades. The *.conf and *_additional.conf files are controlled by FreePBX and can be overwritten. The *_custom.conf files are provided for custom editing and are never overwritten by FreePBX. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Avi Miller wrote: You could, but it'll get overwritten by any FreePBX upgrades. The *.conf and *_additional.conf files are controlled by FreePBX and can be overwritten. I thought I should clarify this statement: I meant that FreePBX could overwrite both the *.conf and the *_additional.conf files. You are strongly advised NOT to edit either of those types of files. All editing should be restricted to the *_custom.conf files. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: Maybe its just me, but it appears its no where near usable even with the latest beta1 code. Its just you. I have FreePBX running on 6 production boxes across the country. I do very little additional scripting. 5 of the servers have a Eicon Diva Server V-4BRI card. The other (head office) server has a Digium TDM400P (4x FXO) and a Sangoma a101u (ISDN20). FreePBX manages all of those lines just fine. What problems are you having? Personally, I don't have any requirements over and above the standard FreePBX installation. And if I do, I just go bug the developers until they put it in. :) cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: address zap interfaces, but implies all four lines have to drop into the same context. Not usable given the above. The new beta (2.1) allows you to route inbound based on Zap channel -- you could set each channel to route to a specific destination, and FreePBX will create the dialplan for you. After implementing the beta1 code yesterday, it looks like they removed several items (such as being able to edit conf files directly, crm, etc) with no indication as to whether that is permanent or what. No, those are [EMAIL PROTECTED] specific additions and have never been part of AMP or FreePBX (i.e. the maintenance tab and the SugarCRM integration). FreePBX is merely the GUI that creates/manages your dialplan. Prior to 2.1 and even post 2.1, I have all my TDM400P inbound calls coming to the same destination: The office IVR. Prior to 2.1, I used the catch-all destination (i.e. no DID/CID defined) for these. Post-2.1, I you could do it by Zap channel. Check out #freepbx on irc.freenode.net for more support, or the Documentation Wiki at http://aussievoip.com.au/wiki/freePBX FreePBX is as flexible as you make it, essentially -- if it doesn't do what you want it to do, feel free to write your own module (or fund the development of one). =D cYa, Avi P.S. I'm not a FreePBX developer -- I just hang out in IRC and bug the real developers periodically. FreePBX does what I need, but obviously Your Mileage May Vary. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: Well... all those things were installed with FreePBX, they just didn't grow there. ;) Honestly, those utilities never been part of FreePBX (nor are they installed by FreePBX). They are only ever installed as part of [EMAIL PROTECTED] However, one of the FreePBX developers is currently implementing a lot of the stuff from [EMAIL PROTECTED] into FreePBX (like the Maintenance tab to hand edit the conf files and the Java SSH client). I've been to the wiki several times, but its very short on any any form of documentation. And, obviously the Handbook was borrowed from the [EMAIL PROTECTED] disto and doesn't actually follow the FreePBX implementations. Obviously, the Wiki documentation is a work-in-progress. Its a lot better than it used to be. If there are specific sections that you'd like more information about, please let the guys in the #freepbx channel know. Is there a user's mailing list for this, or just the irc channel? You can subscribe to the amportal-users list via the SourceForge project for AMP (which is now FreePBX). cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: Actually, they were installed by FreePBX and I still have the iso disk to prove it The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an ISO. FreePBX is simply one of the many software applications that have been combined to form the [EMAIL PROTECTED] distribution. :) I've never implemented [EMAIL PROTECTED], but it does appear that must have been the starting point for FreePBX. Actually, the other way around: FreePBX was probably one of the starting points for [EMAIL PROTECTED] :) Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: So, how do you know which conf files one can hand edit versus those that might be overwritten? You may only change the *_custom.conf files. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] FreePBX in production?
Rich Adamson wrote: zap interface, but apparently undid what existed to edit conf files, crm, etc. That made things look like a step backwards. Yeah, a lot of people get confused about that. I was just trying to clear things up. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] Problems with Eicon Diva V-4BRI - 2nd Port
Hey guys This is probably for Armin, but I thought maybe someone else might have something I could try. I'm having a problem with one of my Eicon Diva V-4BRI cards and I'm trying to work out if its a driver configuration error, card failure or telco problem: I have an Asterisk box running -- the Eicon drivers see all four ports on my card, capi.conf is configured with all four ports and when I issue a capi info on the Asterisk CLI, I see all four ports. The problem is, I only have two ISDN2 lines connected, to ports one and two. I configured capi.conf with only two controllers, but Asterisk still sees all four. If I try to dial-out on anything but the first port, it just hangs (and I get an engaged tone once two lines are up, suggesting that the 2nd port just isn't working properly). However, I've swapped cables and NT1s, and both lines work in Port 1 of the Eicon Diva. But I can't see any configuration difference between this server and the other server I have that is working fine across its two ports. Does anyone know of any Eicon diagnostics I can run to see if the second port is actually up? I've been through the Diva webinterface, but I'm not actually sure what to look for! Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Problems with Eicon Diva V-4BRI - 2nd Port
Avi Miller wrote: This is probably for Armin, but I thought maybe someone else might have something I could try. I'm having a problem with one of my Eicon Diva V-4BRI cards and I'm trying to work out if its a driver configuration error, card failure or telco problem: Replying to myself to let you all know that I'm both a) a moron and b) a genius. Rebuilding the Eicon drivers and checking the configuration for the 1 millionth time revealed the single misconfiguration (set to p2mp instead of p2p) which caused my problems. Glad to report its all working again. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Olivier Krief wrote: When writing receiving a fax over CAPI, do you mean receiving a fax over CAPI with Asterisk and processing it with spandsp ? No, with a full Eicon Diva 4BRI card, it does hardware faxing. Instead of using rxfax (which uses spandsp), you'd use capicommand(receivefax) which does a hardware receive on-board. Also, I can confirm that you can receive faxes using spandsp on the V-4BRI (voice-only) board. Which is nifty. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Question about Asterisk realtime
Tielin Xu wrote: I noticed that there is no ip address stored for my softphone in Mysql, how does the Asterisk know which computer my softphone is running? I checked the config files, no softphone registrations in sip.conf. freePBX stores your phone information in sip_additional.conf and does not use Asterisk Realtime. Asterisk knows what IP address your phone is on once the phone registers with the Asterisk server. To see if your phone is registered, run 'sip show peers' at the Asterisk console and see if Asterisk sees your phone's IP address. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Question about Asterisk realtime
Tielin Xu wrote: I'd like to use FreePBX, it seems some setup inconsistency with Asterisk RealTime, do you know any other good admin tool for Asterisk? FreePBX is not designed to work with Asterisk RealTime. I don't know of a GUI to configure RealTime myself. :) -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Need help with getting EXTEN from pstn hunt group
Jim Freeze wrote: I have a TDM card with 4 lines on a hunt group coming in. The latest version of FreePBX (2.1 Beta 1 - currently in SVN, but should be released soon, I'm told) allows you to create inbound routes based on Zap Channel, which I believe is what you're look for. You may want to grab a copy of 2.1 from SVN to see how they do the Zap-channel based inbound routing in extension.conf and extensions_additional.conf Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Armin Schindler wrote: Using spandsp and V-4BRI does not work? That will work. It's just that the on-board fax capabilities won't work, but any other software fax will work like with other cards. Just a note that I've never managed to get this to work on my V-4BRI cards: If I attempt to use SpanDSP to send or receive a fax, Asterisk will crash. This happens on multiple servers, so now I don't even bother compiling SpanDSP support onto my BRI-only Asterisk servers. If anyone knows how to actually get this working, I'm all ears. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Polycom MWI
Kerry Garrison wrote: Didn't help. Could I be missing something else? My phone.cfg looks like this: mwi msg.mwi.1.subscribe=300 msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97/ And sip.conf for extension 300: [300] username=300 type=friend secret=*** record_out=Adhoc record_in=Adhoc qualify=no port=5060 pickupgroup=1 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup=1 callerid=Polycom IP501 300 allow=alaw allow=g729 Mine works fine, so I hope that helps. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] CAPI Installation Eicon Diva Server
Armin Schindler wrote: The configuration is as easy as with BRI lines. Can you provide more (like your confs and verbose/debug output)? Also (this isn't directed at you Armin, but I found your email to reply off of to maintain the threading), I created a Wiki page over at the freePBX documentation site, explaining how to configure an Eicon Server 4-BRI for freePBX. It may have some tips for you: http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva Feel free to add/remove information. Its a Wiki after all. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] CAPI Installation Eicon Diva Server
[EMAIL PROTECTED] wrote: Asterisk says it has 30 capi channels available, but my mistake may be in configuring the trunks... When I was debugging my Eicon Diva 4-BRI board, I found it useful to play with extensions_custom.conf (in AMP) just to ensure I got the Custom Dial String absolutely correct. According to the latest chan_capi-cm, the Dial String should be: CAPI/id/number/options Where: id = Contr1 or g1 (Controller or Group ID) number = Phone number options = Things like B or b for Early B3 and other things. I have 'b' in my options, but I do admit that I have no idea what early B3 is. :) Hope that helps in some way, Avi P.S. I wrote a quick config page for the 4-BRI for freePBX here: http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva It might have a few things to consider as well. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)
Jim Rice wrote: I only asked this list as a last resort, having already exhausted many other avenues. I even mentioned that it was OT, but have seen numerous postings for phones of all kinds. A thought: I had similar problems with one phone of mine after I power-cycled it during the provisioning process. Its a known issue with the Polycoms that they can become.. confused.. if power-cycled while they're booting. Have you tried booting the phone offline and formatting its filesystem via the Advanced menu? cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)
Jim Rice wrote: Anyone seen this before? I'm not sure about that exact error, but I get these systems if I stuff up the XML in sip.cfg or phone1.cfg (or the specifc phone equivalents). -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] AMP / Maintenance-Button missing
Thomas Broda wrote: Which component do I have to install in order to get the Maintenance setup? The Maintenance tab is part of [EMAIL PROTECTED] and not AMP/freePBX. You'll only see it on an [EMAIL PROTECTED] installation. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials - FIXED!
Noah Miller wrote: Another idea: Can you create the mac address-directory.xml files as symlinks to the central file? Great idea and it works, too! :) Now I just need to make 50 symlinks.. luckily I have a list of mac addresses, so its just a Bash script away. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
Dinesh Nair wrote: the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. I had a similar problem connecting Asterisk to an Avaya IP403 via OOH323: In the end, I removed all the disallow=all and allow=codec lines in Asterisk. This seems to have allowed the two systems to overcome the codec negotiation problems they were having and proceed with actual audio transfer. :) I have no idea if this is related, but I thought I'd just throw that out there, if only for testing purposes. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
Dinesh Nair wrote: more tests reveal that with ohphone, calls from SIP-ohphone work fine with audio passed both ways. however when ohphone calls a SIP device, the call is hungup when the SIP device answers. This was sort of my problem too. I have two Asterisk servers, with an IAX2 trunk between them: Phone - Asterisk 1 - IAX - Asterisk 2 - H323 - Avaya IP403 - Phone If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya, it worked fine. If I dialled from a phone on the Avaya, the SIP phone would ring, but the call would drop as soon as it was answered because of codec negotiation failure. After removing the various disallow= and allow= lines, the codec negotation is now successful in both directions. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Diva Server BRI echo options
Giuseppe wrote: I'm always getting this error when echo cancellation should start. What does your /etc/asterisk/capi.conf look like? Also, have you configured your Eicon correctly? You may need to enable the Eicon web interface and check that each port is correct. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Diva Server BRI echo options
Giuseppe wrote: ntmode=yes ;if isdn card operates in nt mode, set this to This should be set to no -- you should be in TE mode. echotail=64 ;echo cancel tail setting bridge=yes ;native bridging (CAPI line interconnect) if I don't have either of these settings for any of my BRI definitions: [V4BRI-1] isdnmode=DID incomingmsn=* controller=1 softdtmf=0 accountcode= context=from-trunk group=1 callgroup=1 echocancel=yes devices=2 -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials
Mojo with Horan Company, LLC wrote: if you reboot your phones from the asterisk server ie via cron or so, that reboot script could potentially delete the phone-specific directory xml before the sip message is sent Sadly, that doesn't work -- the Polycoms store their directories locally as well and re-upload them on reboot. Though, if you have a sample of that remote reboot script for the phones, I'd appreciate a copy. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Echo cancellation problem
Giuseppe wrote: Can anybody tell me if there is some error or something missing in this configuration please? I have the same card in a few of my servers and the echo canceller works just fine. I'm not 100% sure, but something does jump out at me: == ISDN3: Answering for 'x' -- Playing 'wsa_benvenuto_lib_uni' (language 'it') This plays *before* the echo canceller starts. If you suppress this, does the echo can get a chance to setup successfully? Mar 31 16:40:21 WARNING[30181]: file.c:1029 ast_waitstream: Unexpected control subclass '14' == ISDN3: Setting up echo canceller (PLCI=0x103, function=1, options=4, tail=64) == ISDN3: Setting up DTMF detector (PLCI=0x103, flag=1) -- ISDN3: Error setting up echo canceller (PLCI=0x103) Mar 31 16:40:21 WARNING[29878]: chan_capi.c:3334 show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x103 Command=FACILITY_CONF,0x8497 CAPI INFO 0x300b: Facility not supported cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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] OT: Polycom IP501 and Speed Dials
Hi gang, I know this is off-topic for Asterisk, but I don't know where else to ask: I've setup a central directory.xml file for my Polycom IP501 phones with a list of all the internal extensions. None of them have sd1/sd as I don't want to enable any speed dials, just have a list in each phone. However, when a phone boots, it seems to pick a random entry and put it on the second line key as a speed dial entry! Anyone have any idea why and how to stop it? Also, could someone confirm that once a phone loads the default directory, it then maintains its own copy? So if I want to change the directory from the FTP server, I have to edit every single phone-specific XML file, or will the phone overwrite that on reboot? Essentially, I'm looking for a way to manage the directory from a central location. Thanks, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Reporting?
Doug Lytle wrote: shameless plug Something like this perhaps? VERY cool! I agree. When does that get released? :) -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
Re: [Asterisk-Users] Re: AAH lost my IVR phrases
Jim Hanlon wrote: 1. The alterations to the config files made via AMP Setup pages are archived in the Asterisk DBMS, but changes made via the AMP Maintenance pages are not (Apparently. It's hard to be sure what the rules are). This is an [EMAIL PROTECTED] issue: The Setup page is provided by AMP (now called freePBX, btw), but the Maintenance page is NOT. So, the [EMAIL PROTECTED] system allows you to change configuration files built by AMP, which is where the confusion comes in. If you install Asterisk and AMP/freePBX manually, there is no maintenance tab, so there is less opportunity for you to overwrite the pre-baked configuration files. :) The rules are fairly straightforward though: Anything *_additional.conf is written by AMP/freePBX and should not be touched. Anything *_custom.conf is never touched by AMP and can be manually edited. Anything *.conf is only overwritten on upgrades of AMP, so you should take care if you edit those files. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.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
