Re: [Asterisk-Users] hardware question
what we needed and I want to ask if I understand the naming correctly: FXS = pstn-signals for calling someone (towards central pbx/server) and knowing that someone is calling you FXO = ...? FXS has a phone plugged in it FXO hgas a phone line plugged in it http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html http://asteriskdocs.org ATA = connecting potstelephone to lan/computer/... other protocol Yes What I don't understand is that ATA-devices which only have an FXS-port are also able to recieve calls or not? So, what is the difference with an FXO-interface? Read the specs for the sipura 3000 for example at their site So, since for now we'll only need to be able to talk to one pots line (and one bri-line), we'll need one FXS-interface (to recieve siand you need 1 FXO I've had a look at the digium hardware-store since I wanted to be sur eit will work with asterisk. For pstn, I suppose a TDM400P with one FXS and one FXO module(?). I also saw the Asterisk Developpers kit PCI which fullfils these requirements, which states that it is only for Asterisk-developpers. Anyone can buy one, it just was a good bundle for test. I saw that on the page of the Asterisk Developpers kit PCI it states that one needs linux 2.4. I have asterisk 1.0.6 installed on this computer and it seems to have drivers for 2.6. So, the question: does the hardware works with linux 2.6.x-kernels? It definitely works, but there are some issues that someone who has done it may be able to discuss. Otherwise, search the mailing list using google: bri card site:lists.digium.com 2.6 kernel site:lists.digium.com and any other search expressions you can think of. Do the same without the site: parameter too, replacing it with asterisk. googling asterisk bri card brings up a lot of interesting stuff as does asterisk 2.6 hth ___ 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] Thank you for the timely suggestion
I have been searching for the necessary components for my setup from sometime back; yet to install Asterisk and will be installing softphones on Linux Server and on all windows PCs(most of them are Windows Xp,others are Windows 2000 professional,Windows 98); but could not decide which softphone to use still searching for the softphone.. Discover Yahoo! Stay in touch with email, IM, photo sharing & more. Check it out!___ 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] IP PHONE iareaphone x100, tested??
so i was looking at the internet and i read a lot, the cheapest are the Grandstream BudgetTone but some reviews of this list says they are not so good ... so i found Many people hate these phones, yet I've found my 3 BT100 to be excellent for a network of friends and associates (not everyday office for example). iareaphones but i can't find reviews about them, i would like to know if someone has experience with them, at their site the phone seems to be done to work for Asterisk ... but im not gonna buy something without a good review There is at least one review of the older AT320 on the wiki that covers some of the same ground. I own two of these and one Netweb 120, and all work pretty well with the latest software. See also: http://www.voip-info.org/wiki-NetWebGroup You can buy from iareaphones, I placed a couple of orders with them (they sell Polycom as well when you're ready for a real phone) and they did fine with customer service. These IAX phones are more flexible than most because they can be programmed via dialpad, web interface and a windows utility called palmtool. You have easy access to local dialplan (which I don't use with asterisk) and digitmap, which I do use. hth ___ 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] bypass incoming ring..is it possible?
Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Try setting callerid=no on the FXO channel ___ 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] Thank you for the timely suggestion
Hi, try xlite if you have enough bandwitdh for G711 codec requirement.. try firefly if you want to use G729 codec freely (linked via dll).. both of them are the best freeware softphone for windows. Best regards, Stevanus infra struct wrote: I have been searching for the necessary components for my setup from sometime back; yet to install Asterisk and will be installing softphones on Linux Server and on all windows PCs(most of them are Windows Xp,others are Windows 2000 professional,Windows 98); but could not decide which softphone to use still searching for the softphone.. Discover Yahoo! Stay in touch with email, IM, photo sharing more. Check it out! ___ 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
[Asterisk-Users] Asterisk Engineer/Programmer required
Hi, Were looking for an experienced Asterisk engineer/programmer to configure and install Asterisk systems. This is a full time position and the person will be based in Asia. Share options are available and we are open to negotiate. Minimum of 1 year experience installing and configuring Asterisk systems. Interested parties, please email your resume and expected salary to : info @ motavi . com Thank you! ___ 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] GXP2000 and hint LED's
Bugger, thanks for replying and telling me, might send a request through to Grandstream and see when they intend on releasing it. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Thursday, 9 June 2005 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP2000 and hint LED's On Thu, 9 Jun 2005, James Bean wrote: Has anyone got the hint function working, and maybe with the GXP2000. I don't think the current firmware release for the GXP-2000 supports SUBSCRIBE/NOTIFY. That functionality is to be released at a later date. Peter ___ 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
[Asterisk-Users] TDM400P strangeness
Hi List, I have a test asterisk box with a TDM400P with 4 FXO modules plugged in. Yesterday I could use the box without any issues - no problems. This morning, the sound on the box was absolutely horrible. After some fiddling about, I have rebooted the box, and now asterisk refuses to start! Here's the message I get: Jun 9 10:45:53 WARNING[3297]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address Jun 9 10:45:53 ERROR[3297]: chan_zap.c:6201 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 9 10:45:53 ERROR[3297]: chan_zap.c:9148 setup_zap: Unable to register channel '1-4' Jun 9 10:45:53 WARNING[3297]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Jun 9 10:45:53 WARNING[3297]: loader.c:440 load_modules: Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! [EMAIL PROTECTED]:~# Ouch ... error while writing audio data: : Broken pipe Any ideas? Help! Cheers, Jean-Michel ___ 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] bypass incoming ring..is it possible?
Hi, I've tried your suggestion but the result is still the same... Have another suggestion? Best regards, Stevanus Wilson Pickett wrote: Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Try setting callerid=no on the FXO channel ___ 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
Re: [Asterisk-Users] hardware question
Hi Michel, Michel Brabants wrote: I didn't see a bri-adapter on the digium-site, only pri it seems. Any recommendation on that or is there a bri-adapter from digium. I'm also open to other vendors. I saw that there are others which ar ecompatible with asterisk, but I don't have a lot of time and want to be sure that it works with asterisk. My personal recommendations are: - For a one-port BRI-Card: choose one with a HFC-S chip (e. g. Acer ISDN 128 Surf PCI) - For multi-port BRI: have a look at www.beronet.com and www.junghanns.net You can use all this cards with bristuff from www.junghanns.net I have had a look at the mISDN-hardwarepage. I suppose I can choose any bri-card that is supported by them? Is mISDN supported by asterisk? There is a chan_mISDN driver from beronet. But as far as I can tell, it is not very stable. I use bristuff for all our BRI cards - no problems. Regards, Henry ___ 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] performance of * in several scenarios
Nobody ? :-( -b - Original Message - From: barney To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, June 08, 2005 11:39 AM Subject: [Asterisk-Users] performance of * in several scenarios Hi, Is here someone who could provide meany information from practical using of * ? I need to know more about performance. The main question is: "How many extensions should i have configuredin and provided with my * box in several cases": 1. * is usedonly for SIP signalling, no rtp stream is going through * (always using reinvite and no nat used in lan/wan) 2. * is used for signalling, rtp stream is going between UAs, but rtp stream is going through *, when is routed outside (from SIP to TDM world) via SIP trunk 3. * is used for signalling, rtp stream is going between UAs, but rtp stream is going through *, when is routed outside (from SIP to TDM world) via local CAPI/ZAP interface 4. * is used for signalling, rtp stream is always going through (never reinvite) Common details: - no codec translations / only one codec used in whole network - voicemail system and other services like wakeup calls, weather information, ... are running on other (dedicated) * box (hope that its possible) - no frontend like SER used Are there any tables or some tools, which could make some calculations for me ? Thanks, B ___ 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] Notice Message
hi all i have a notice message that comming frequently says that pbc.c:1329 pbx_extention_helper; cannot find extention context 'default' anyone know this warning and how to solve because its realy anoying thks roy __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Re: mISDN + chan_misdn.so + winbond issue
Hi, This is to let you all know that I have it working now. Thanks to Titus who supplied his list of a working combination ( http://amatisoft.homelinux.com/demo/index.html ) and some other tips. For archive and history purposes I will post my combination which may help others who will run into this: Packages: * kernel 2.6.9 * mISDN kernel patch from PBX4Linux (http://isdn.jolly.de/download/v3.0beta/mISDN_for_PBX4Linux_2005_03_06.tar.gz) * mISDN user from PBX4Linux (http://isdn.jolly.de/download/v3.0beta/mISDNuser_for_PBX4Linux_2005_01_28.tar.gz) * chan_misdn 0.1.0 * asterisk 1.0.7 Important factor was also to have the correct 'layermask' parameter when loading the winbond module. This had to be 0xf and not 0x1 , I am still thankful to Titus who pointed this out, at least I could not find any documentation on this parameter but it turned out to be an important one. My modprobe looks now like this: modprobe w6692pci protocol=2 layermask=0xf Best regards, Michel Koenen On 6/6/05, Michel Koenen [EMAIL PROTECTED] wrote: Hi all, Does anybody of you have the winbond w6692 working with the mISDN/chan_misdn.so? When loading chan_misdn.so from Asterisk, I get a No lower Id port:1 error. The /var/log/messages file says: MISDN free_device: entitylist not empty I'm using Linux 2.6.11.11 + mISDN-CVS-2005-05-01 + Asterisk 1.0.7 + Zaptel 1.0.7 chan_misdn build from chan_misdn-beta-0.0.3-rc6 and against mISDNuser-CVS-2004-08-29. The /dev/mISDN node was also created. I'm loading the kernel modules this way: modprobe zaptel modprobe ztdummy modprobe mISDN_core modprobe mISDN_l1 modprobe mISDN_l2 modprobe l3udss1 modprobe mISDN_dsp modprobe w6692pci protocol=2 layermask=1 Then I start asterisk: asterisk -c -vv -dd When loading chan_misdn.so , Asterisk complains and exits after the last error line below [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 No lower Id port:1 init_stack: No such file or directory Contents of the /var/log/messages for all above commands: Jun 5 20:25:20 pbx kernel: Zapata Telephony Interface Registered on major 196 Jun 5 20:25:25 pbx kernel: Registered tone zone 0 (United States / North America) Jun 5 20:25:48 pbx kernel: Modular ISDN Stack core $Revision: 1.25 $ Jun 5 20:25:53 pbx kernel: ISDN L1 driver version 1.11 Jun 5 20:25:56 pbx kernel: ISDN L2 driver version 1.20 Jun 5 20:26:02 pbx kernel: mISDN: DSS1 Rev. 1.29 Jun 5 20:26:07 pbx kernel: mISDN_dsp: Audio DSP Rev. 1.10 (debug=0x0) Jun 5 20:26:20 pbx kernel: Winbond W6692 PCI driver Rev. 1.13 Jun 5 20:26:21 pbx kernel: PCI: Found IRQ 9 for device :00:0f.0 Jun 5 20:26:21 pbx kernel: mISDN_w6692: found adapter Winbond W6692 at :00:0f.0 Jun 5 20:26:21 pbx kernel: W6692: Winbond W6692 version (0): W6692 V00 Jun 5 20:26:21 pbx kernel: w6692: IRQ 9 count 4 Jun 5 20:26:21 pbx kernel: w6692 1 cards installed Jun 5 20:26:34 pbx kernel: MISDN free_device: entitylist not empty Am I using wrong or incompatible source versions or is this a bug or am I doing something wrong? Btw the misdn.conf contains: [general] language=en immediate=no debug=0 [mycard] context=incoming ports=1,2 msns=72 Using ports=1 or ports=2 or changing msns gives the same problems.. When you have a working configuration, I am curious which source versions of needed packages you have used. Thank you in advance for your response. Best regards, Michel Koenen ___ 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] TDM400P strangeness
Jean-Michel Hiver a écrit : Hi List, I have a test asterisk box with a TDM400P with 4 FXO modules plugged in. Yesterday I could use the box without any issues - no problems. This morning, the sound on the box was absolutely horrible. After some fiddling about, I have rebooted the box, and now asterisk refuses to start! Here's the message I get: Jun 9 10:45:53 WARNING[3297]: chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address Jun 9 10:45:53 ERROR[3297]: chan_zap.c:6201 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jun 9 10:45:53 ERROR[3297]: chan_zap.c:9148 setup_zap: Unable to register channel '1-4' Jun 9 10:45:53 WARNING[3297]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Jun 9 10:45:53 WARNING[3297]: loader.c:440 load_modules: Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! [EMAIL PROTECTED]:~# Ouch ... error while writing audio data: : Broken pipe Looks like kernel module is not loaded or TDM not initialized modprobe wctdm ztcfg Thanks, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ 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] TDM400P... ignoring hanguponpolarityswitch
I've made a backport of this patch for asterisk stable. You can get it here: http://www.maxosystem.net/asterisk . The page is in Spanish, but you just need to download and apply the patch to chan_zap.c. It also works with bristuff patch applied. Julian J. M. On 6/9/05, Neil and Fiona [EMAIL PROTECTED] wrote: Is there a list of options that are valid for stable? I downgraded from Head to stable when I had IAX trunking problems (one way audio) with a VSP. So I am using my conf files from Head, which could be the problem. I've got a copy of sample config files from 1.07 (Or I think they are, I didn't label it well when I archived it). It seems to have the option in it. There has been a patch in Head for the IAX2 trunking problem, so I think I could go back to head. ___ 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] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?
Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ 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] New version 1.013 of Asterisk VConfig
This is mostly a testing/bug fix release. Hopefully by the next version I will have some real documentation up on the site. Since it's primarily a platform rather than an end user system, without documentation it's not nearly as useful as it could be. http://asterisk.ochsnet.com Chris ___ 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] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?
You're connected to a p2mp bri, switch to bri_cpe_p2mp Matteo. Il giorno mer, 08-06-2005 alle 19:54 +0200, Robert Rozman ha scritto: Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ 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
[Asterisk-Users] Softphone for Linux desktops
Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do Thanks ___ 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] TDM04B
Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks. ___ 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] Softphone for Linux desktops
Hi, try xlite, it has linux version.. Best regards, Stevanus Eric Bishop wrote: Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do Thanks ___ 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
RE: [Asterisk-Users] Softphone for Linux desktops
Hi, -Original Message- We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do For SIP I love kphone. Nice interface, works simple enough, available in many popular linux distro's. For IAX I'd go with iaxComm. Florian ___ 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] Do I need a ring capacitor to use TDM400P cards in UK
On Wed, 2005-06-08 at 23:39 +0100, David John Walsh wrote: Angus Jumping in with both feet a BT socket with a capacitor in is commonly refered to as a Master socket, and are very cheap even without wholesale. It gets its name from being the socket that BT installed into the house for the line, all other sockets in the house will be slave or secondary (ie no capacitor) (and its against the law to play with the one BT installed - but thats off topic!) ..and it complicated my understanding of how to get ADSL working at the same time so ADSL filter was installed before the Master... UK Phones at homes historically had a separate bell - mounted in the hallway. Phones where then placed where convenient. This allowed one loud bell (sucking current) and multiple (quieter, less current thirsty) phones... To do this, the 2-wire line from the Telco was altered into a three wire line inside the residence, the job of the 'Master Jack'. This is done with a capacitor from one of the legs to provide the third wire. Look inside the 'Master' to confirm... (there might also be a resistor from the other leg to the new third leg too). (I can remember playing with a crystal radio set, that needed an earth, and the instructions saying to use the metal (silver coloured) finger stop on the rotary dial as an earth - so there may be an earth as a fourth wire...) Because of this - many phones sold in the UK will only ring via this third wire... I vaguely remember bringing a cordless phone from the UK to South Africa (where the US 2-wire equipment work fine) and adding a capacitor inside the phone to make it Ring... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Clicks in audio with TE100P PRI
On Thursday 09 June 2005 00:52, James Bean wrote: span=1,1,0,ccs,hdb3 The same thing happens. Did you rerun ztcfg? I have heard rumour (but not seen it myself) that you need to fully reset (power off/on, not just reboot) to get the card to accept a new clocking method. You may consider also that if I connect PAP2 to LAN everything works, also if I use other ip phone from internet works fine. This tells me that it's not related to LAN. I also check if I'm loosing interrupts and everything seems ok. Also I pull out the TDM400 from the box. This tells me it's got nothing to do with the TDM400 or lost interrupts. At last I change jitterbuffer=16 and it works better, the clicks are reduced. Could this be possible? What is the function of this parameter in zapata.conf? You're just masking the issue. I should tell you that the TE100P is connected to another E1 board (not a live E1) from Natural Microsystems which acts as a gateway to PSTN. This board works as a PRI master but I don't think that this could be the problem as long as using other phones or in LAN it works perfectly and the voice is clear with no clicks o sound looses. Please try what I suggested above. I am confident it'll solve your problem. -A. ___ 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] TDM04B
Is it true that a FXO port will NOT provide a dial tone? On Wed Jun 08, 2005 at 08:23:54PM +0300, [EMAIL PROTECTED] wrote: Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks. ___ 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
Re: [Asterisk-Users] TDM04B
On Thursday 09 June 2005 06:32, [EMAIL PROTECTED] wrote: Is it true that a FXO port will NOT provide a dial tone? FXO means it connects to a central Office -- it accepts dialtone and ring (it acts as a telephone) FXS means it connects to a Station (telephone) -- it provides dialtone and ring (it acts as a telephone network) Easy to remember: FXO connects to an Office, FXS connects to a Station or Set. You ordered a TDM04B which is four FXO ports, which is for connecting four telephone lines to. I *think* you want a TDM40B which is four FXS ports, which is for connecting four telephones to. I'm *positive* that Digium (or your reseller) will swap this out for no extra charge unless there's a restocking fee for having to order it in. -A. ___ 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] 180 Ringing? (BUG?)
Voilà. Now i know where is the problem. I use 2 ISDN channels with a with a fritz! card and the junghanns capi drivers. The problem appears with SIP to ISDN calls. The SIP 180 ringing message doesn't appear because the ISDN PBX sends the ALERT message in-band (channel B), and not in the D channel. So Asterisk doesn't know when the ISDN channel is ringing. With my configuration Asterisk can not understand the in-band signalling for the capi channels, is it possible to use in-band signallisation for capi channels? Mirko ___ 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] More than one account from the same provider?
Inbound is the problem - I am regisitering to the host and receiving faxes. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Wednesday, June 08, 2005 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] More than one account from the same provider? On Jun 8, 2005, at 6:15 PM, Chris Mason (Lists) wrote: I have had good success with my efforts to send faxes over voip using ulaw, surprisingly, and I want to move it from testing to reality. I have an account with Teliax, who have been very good. For voice I use g729 and ulaw, but for faxing I can only allow ulaw. However, Teliax only sets the codec preferences by account. I have another account, but I can't see a way to register two accounts with one server. Any ideas? Chris Mason Outbound or Inbound? If outbound (you said SENDing faxes above, so I'm guessing here) you're not registering, you're connecting via the HOST, USERNAME and SECRET in the context in IAX.conf, right? Robert Goodyear Brand Up LLC http://www.brand-up.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 ___ 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] TDM04B
I'm *positive* that Digium (or your reseller) will swap this out for no extra charge unless there's a restocking fee for having to order it in. Actually, last time I looked there wxas a difference in price. Weren't the FXO a little more expensive? ___ 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] Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using for the last several months:
[teliax]; for incoming calls
context=teliax-incoming
type=user
auth=md5
secret=mymd5secret
disallow=all
allow=gsm
trunk=no
[teliaxout] ; for outgoing calls
type=peer
host=voip.teliax.com
username=myname
auth=md5
secret=mymd5secret ; provided by teliax
disallow=all
allow=gsm
trunk=no
Calls are then placed using something like:
; Calls directed to Teliax.com
; long distance calls completed via Teliax.com
exten = _1NX,1,SetCallerID(3035551212|a)
exten = _1NX,2,SetCIDName(MyName|a)
exten = _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
exten = _1NX,4,Congestion
If I recall correctly, the majority of the above was provided in an
email from teliax when I signed up.
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[Asterisk-Users] Asterisk to Cisco Unity
Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ 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] * @ Home: All Circuits busy
i am a newbie, but have you tried genzaptelconf -s -d Dan On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: All, I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make internal IP calls with no problems, but when I try to dial out I get a message that All Circuits are Busy. I looked into the Zapata.conf files and such but see no modifications. Is there a step that I am missing?? Does anyone have documentation of step-by-step config for this TDM40B card? Thanks, Marc ___ 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
[Asterisk-Users] Pickup problem
Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib ___ 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] Clicks in audio with TE100P PRI
On Thu, 9 Jun 2005, Andrew Kohlsmith wrote: I also check if I'm loosing interrupts and everything seems ok. Also I pull out the TDM400 from the box. This tells me it's got nothing to do with the TDM400 or lost interrupts. It could be that the user-land side (i.e. Asterisk as opposed to Zaptel) does not run often enough. A similar issue went away once we tuned on the real time scheduling for the Asterisk process. Peter ___ 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] More than one account from the same provider?
I have had good success with my efforts to send faxes over voip using ulaw, surprisingly, and I want to move it from testing to reality. I have an account with Teliax, who have been very good. For voice I use g729 and ulaw, but for faxing I can only allow ulaw. However, Teliax only sets the codec preferences by account. I have another account, but I can't see a way to register two accounts with one server. Any ideas? If they are truly two separate accounts with two register statements (and two userid/passwords), I would guess that two different incoming outgoing contexts would work with iax. (SIP accounts will probably not work per Olle's recent post where incoming calls match IP address and essentially ignores userid/passwords.) If you are using iax, then within each context you can specify the codec, something like: disallow=all allow=gsm ;ilbc Just recently I played around with changing codec's on my teliax account. Their Account page provides you with the option to click on IAX and then select the codec, or, click on SIP and select the codec. However, that page is not very clear that you must click on either IAX or SIP before selecting the codec. I've basically selected all codecs that I support, and then in the iax.conf included the entries shown above. Obviously, you can tell that I've been playing with ilbc given the commented statement. Its been working fine. Rich ___ 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] TDM04B
[EMAIL PROTECTED] wrote: Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. This board is FXO which you plug incoming phone lines into it. So plugging in a handset unless it's a butt set it will not give you any dial tone. In fact you damage the port doing this to it. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks. ___ 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
Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch
Thanks Julian, I tried installing cvs-head today but it crashed on compile and the machine rebooted when I did make clean. I'll try the patch and see how I go. On Thu, 2005-06-09 at 08:55 +0100, Julian J. M. wrote: I've made a backport of this patch for asterisk stable. You can get it here: http://www.maxosystem.net/asterisk . The page is in Spanish, but you just need to download and apply the patch to chan_zap.c. It also works with bristuff patch applied. Julian J. M. On 6/9/05, Neil and Fiona [EMAIL PROTECTED] wrote: Is there a list of options that are valid for stable? I downgraded from Head to stable when I had IAX trunking problems (one way audio) with a VSP. So I am using my conf files from Head, which could be the problem. I've got a copy of sample config files from 1.07 (Or I think they are, I didn't label it well when I archived it). It seems to have the option in it. There has been a patch in Head for the IAX2 trunking problem, so I think I could go back to head. ___ 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] OT: Please comment on Dvorak's troll
On 6/7/05, Michael Graves [EMAIL PROTECTED] wrote: On Mon, 6 Jun 2005 11:17:20 -0600, Colin Anderson wrote: http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for free VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of truth in what he's saying. ILEC's would be crazy to not consider this kind of lock in, since it's pretty obvious that packet voice networks are going to supplant circuit networks completely in, say, 20 years. Maybe sooner. Actually, Bob Cringley, another pundit found on the PBS web site raised this matter a few weeks ago. I suspect that IAX2 with some encryption could port hop around and not be easily tracked as VOIP traffic. But in any case there has to be some regulatory stance on what is permitted over a network. Certainly there are non-telco carriers like Covad, whom I use, that would not concern themselves about the nature of the traffic. Michael -- Actually, the FCC has already come down hard on an independent phone company that blocked VoIP traffic for a number of years. Vonage complained, and finally won: http://www.pcpro.co.uk/news/70081/us-slaps-fine-on-company-blocking-voip.html The telcos have seen this coming for years, and many of them are getting into the Video over DSL space as a means to compete going forward. Off topic ___ 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] TDM04B
I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Assuming you have the part number correct, the TDM04B is meant to connect to analog pstn lines, not to telephones. If you want a TDM card to connect to analog telephones, the part number is TDM40B. ___ 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] TDM04B
Thanks. Seems I misunderstood quite a bit. On Thu Jun 09, 2005 at 07:43:11AM -0600, Rich Adamson wrote: I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Assuming you have the part number correct, the TDM04B is meant to connect to analog pstn lines, not to telephones. If you want a TDM card to connect to analog telephones, the part number is TDM40B. ___ 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
[Asterisk-Users] 3COM NBX SuperStack 3
Hi, I've looked pretty wide on Google for this so I don't think it's been asked before. Has anyone had experience integrating Asterisk with a 3COM NBX system? The only way that I can see that looks possible is via the 3Com NBX ConneXtions H.323 product and then one of the H.323 Asterisk channels. Has anyone done this? Any pointers as to what to avoid or whether to even attempt it? I would much prefer a SIP solution to the NBX but this does not appear to exist. Has anyone found such a solution? Thanks in advance Martin ___ 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] * @ Home: All Circuits busy
I got these errors and my hardware is working so I do not think they are an issue Hint: insmod errors Removing zaptel module: zaptel: Device or resource busy What about the ztdummy module? Dan On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dean, Here are the results of the genzaptelconf -s -d. As you can see, it is throwing some errors, but I am a bit of a newbie so any help you could provide would be greatly appreciated! [EMAIL PROTECTED] /]# genzaptelconf -s -d STOPPING ASTERISK Asterisk ended with exit status 0 Asterisk shutdown normally. Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Unloading zaptel hardware drivers: Removing zaptel module: zaptel: Device or resource busy [FAILED] Loading zaptel framework: [ OK ] Waiting for zap to come online ...OK Loading zaptel hardware modules: Running ztcfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started ** SIP/200 in position 2 ** SIP/201 in position 3 ** SIP/202 in position 4 Chan Extension Context Language MusicOnHold pseudofrom-pstn en Verbosity is at least 3 [EMAIL PROTECTED] /]# [EMAIL PROTECTED] /]# Thanks, Marc ___ 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
RE: [Asterisk-Users] More than one account from the same provider?
If they are truly two separate accounts with two register statements (and two userid/passwords), I would guess that two different incoming outgoing contexts would work with iax. They are two separate accounts, but the teliax server will always authenticate itself with the username teliax, as set in the stanza below. [teliax] type=friend host=voip.teliax.com auth=md5 secret=test disallow=all allow=ulaw context=teliax-in Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] * @ Home: All Circuits busy
Getting the ztdummy module out of the mix corrected the problems I was having. Try the following: genzaptelconf -s -d After that has completed, with lines attached, I did a rebuild_zaptel Your mileage may vary, but this seemed to do the trick for me. - Original Message - From: Dan Littlejohn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 08, 2005 3:42 PM Subject: Re: [Asterisk-Users] * @ Home: All Circuits busy I got these errors and my hardware is working so I do not think they are an issue Hint: insmod errors Removing zaptel module: zaptel: Device or resource busy What about the ztdummy module? Dan On 6/8/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Dean, Here are the results of the genzaptelconf -s -d. As you can see, it is throwing some errors, but I am a bit of a newbie so any help you could provide would be greatly appreciated! [EMAIL PROTECTED] /]# genzaptelconf -s -d STOPPING ASTERISK Asterisk ended with exit status 0 Asterisk shutdown normally. Disconnected from Asterisk server Asterisk Stopped STOPPING FOP SERVER FOP Server Stopped Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Unloading zaptel hardware drivers: Removing zaptel module: zaptel: Device or resource busy [FAILED] Loading zaptel framework: [ OK ] Waiting for zap to come online ...OK Loading zaptel hardware modules: Running ztcfg: [ OK ] SETTING FILE PERMISSIONS Permissions OK STARTING ASTERISK Asterisk Started STARTING FOP SERVER FOP Server Started ** SIP/200 in position 2 ** SIP/201 in position 3 ** SIP/202 in position 4 Chan Extension Context Language MusicOnHold pseudofrom-pstn en Verbosity is at least 3 [EMAIL PROTECTED] /]# [EMAIL PROTECTED] /]# Thanks, Marc ___ 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 ___ 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] format g729 and Voxee.com
I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used? BTW, I have disallow=all and allow only the codecs that I want to use in both iax.conf and sip.conf. Best Regards, Todd Reese -- Executing SetCallerID(SIP/201-fbb8, 6788896066) in new stack -- Executing Dial(SIP/201-fbb8, IAX2/134:[EMAIL PROTECTED]/17702561571) in new stack -- Called 134:[EMAIL PROTECTED]/17702561571 -- Call accepted by 66.246.246.52 (format g729) -- Format for call is g729 Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:42 NOTICE[6073]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) -- IAX2/66.246.246.52:4569-7 answered SIP/201-fbb8 Jun 8 18:48:51 WARNING[6405]: channel.c:2308 ast_channel_make_compatible: No path to translate from SIP/201-fbb8(2) to IAX2/66.246.246.52:4569-7(256) Jun 8 18:48:51 WARNING[6405]: app_dial.c:1324 dial_exec_full: Had to drop call because I couldn't make SIP/201-fbb8 compatible with IAX2/66.246.246.52:4569-7 -- Hungup 'IAX2/66.246.246.52:4569-7' == Spawn extension (local-access, 17702561571, 2) exited non-zero on 'SIP/201-fbb8' The above implies that Voxee.com is configured for g729 only. I don't use this itsp, but you might check their web site or call them to see what codec options are available. The flip side is go order and install g729 from digium. ___ 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] More than one account from the same provider?
If they are truly two separate accounts with two register statements (and two userid/passwords), I would guess that two different incoming outgoing contexts would work with iax. They are two separate accounts, but the teliax server will always authenticate itself with the username teliax, as set in the stanza below. [teliax] type=friend host=voip.teliax.com auth=md5 secret=test disallow=all allow=ulaw context=teliax-in Try this... get rid of the friend and use the peer and user defs. Then pay close attention to which parameters apply to those two defs. Change the [teliax] to something different, like [teliax-in] and [teliax-out]. I don't remember you mentioning this, but are you using sip or iax? Rich ___ 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] [ADMIN]: subscription failure
Did you go to the web page that is listed at the bottom of every message? Look at the bottom of that web page for the address of the list admin. By the way, that admin address is pretty much standard for ALL mailing lists. On Wed, Jun 08, 2005 at 07:37:35PM -0700, David Koski said: Would an admin please contact me off list? I tried to subscribe from another address and it failed--I got no email to confirm the subscription. I would rather use the other address and need to know if there is a problem with my mail server. ___ 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] format g729 and Voxee.com
Voxee will not accept any calls that are not in G729. You need G729 codec on your Asterisk. Period. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, June 09, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Todd Reese Subject: Re: [Asterisk-Users] format g729 and Voxee.com I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used? BTW, I have disallow=all and allow only the codecs that I want to use in both iax.conf and sip.conf. Best Regards, Todd Reese NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Play MP3 during Record
On 6/9/05, Phuong Nguyen [EMAIL PROTECTED] wrote: Hi all, Does Asterisk support multi thread? I mean: Is it possible to do one of the 2 following scenarios: 1. Play a low background music when the user record his/her voice I don't know why you would want to do that, but here is a hack. Throw two calls into a meetme. One with chan/local playing the mp3, and the other your call that you want to record. Monitor the second leg out and there you go. This would take some tweeking with MeetMe flags but totally possible. Just a hack, there is probably a better way. TIMTOWTDI -Brian ___ 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] More than one account from the same provider?
Try this... get rid of the friend and use the peer and user defs. Then pay close attention to which parameters apply to those two defs. Change the [teliax] to something different, like [teliax-in] and [teliax-out]. All that will do is separate the configs for incoming and outgoing, I need two completely separate incoming and outgoing accounts. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] format g729 and Voxee.com
Kanuri, That must be why, on their setup page (in the members section of their site), they list ulaw, alaw, g.729, gsm and ilbc as "Supported Codecs" ;-) We've used them with ulaw, recently :-) Kanuri, Seshu (Company IT) wrote: Voxee will not accept any calls that are not in G729. You need G729 codec on your Asterisk. Period. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rich Adamson Sent: Thursday, June 09, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Todd Reese Subject: Re: [Asterisk-Users] format g729 and Voxee.com I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used? BTW, I have disallow=all and allow only the codecs that I want to use in both iax.conf and sip.conf. Best Regards, Todd Reese NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 -- Cheers, Mattt. VoIP made easy - http://voip.abovenetworks.net Convergent network specialists - http://abovenetworks.net I have an inferiority complex, but it's not a very good one... ___ 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] Re: format g729 and Voxee.com
Hi everybody, Just to clarify, voxee supports the codecs G729, ulaw, alaw, gsm, ilbc. You will need to force select a particulat codec out of those if you want to use it. We purchased the g729 codecs directly from Digium so it should have no problems working with your * installation if you also purchase it from them. Cheers On 6/9/05, Mattt [EMAIL PROTECTED] wrote: Kanuri, That must be why, on their setup page (in the members section of their site), they list ulaw, alaw, g.729, gsm and ilbc as Supported Codecs ;-) We've used them with ulaw, recently :-) Kanuri, Seshu (Company IT) wrote: Voxee will not accept any calls that are not in G729. You need G729 codec on your Asterisk. Period. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, June 09, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Todd Reese Subject: Re: [Asterisk-Users] format g729 and Voxee.com I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used? BTW, I have disallow=all and allow only the codecs that I want to use in both iax.conf and sip.conf. Best Regards, Todd Reese NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 -- Cheers, Mattt. VoIP made easy - http://voip.abovenetworks.net Convergent network specialists - http://abovenetworks.net I have an inferiority complex, but it's not a very good one... ___ 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] Softphone for Linux desktops
Eric Bishop wrote: We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do Check out Kiax. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ 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] Lingo(.com) and Asterisk
Hello, A long Google search didn't turn any clear answer. Does somebody use Asterisk in combination with Lingo? Thank you, Bas ___ 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] 3COM NBX SuperStack 3
Martin Croome wrote: Hi, I've looked pretty wide on Google for this so I don't think it's been asked before. Has anyone had experience integrating Asterisk with a 3COM NBX system? The only way that I can see that looks possible is via the 3Com NBX ConneXtions H.323 product and then one of the H.323 Asterisk channels. Has anyone done this? Any pointers as to what to avoid or whether to even attempt it? I would much prefer a SIP solution to the NBX but this does not appear to exist. Has anyone found such a solution? Thanks in advance Martin The 3Com NBX also supports analogue loop start (FXS), T1/PRI, E1/PRI, ISDN BRI-ST and Q.SIG/Q.931. You would need the appropriate cards for the NBX and your Asterisk server. I have heard that SIP is planned for future releases of the NBX. Regards -- Chris Hills IT Services North East Worcestershire College ___ 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] 180 Ringing? (BUG?)
I guess that's Early Media Connect, i.e., if the phone supports that (not all do), the channels get bridged just after dial completed, (SIP 183), and what you hear is the remote ring tones (from your telco), not locally generated (as if it received SIP 180 Ringing). What IP phones are you using? You may try xlite and check if you hear ringing tones. Julian. On 6/9/05, Mirko Marghitola [EMAIL PROTECTED] wrote: Voilà. Now i know where is the problem. I use 2 ISDN channels with a with a fritz! card and the junghanns capi drivers. The problem appears with SIP to ISDN calls. The SIP 180 ringing message doesn't appear because the ISDN PBX sends the ALERT message in-band (channel B), and not in the D channel. So Asterisk doesn't know when the ISDN channel is ringing. With my configuration Asterisk can not understand the in-band signalling for the capi channels, is it possible to use in-band signallisation for capi channels? Mirko ___ 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] GXP2000 and hint LED's
I've just checked the download page, and the latest firmware available is 1.0.1.8. Where did you find 1.0.1.9? This phone has some nasty bugs, one of them being that the other end HEARS you after you press the Transfer button and you hear a dialtone. It doesn't send any message to asterisk so that it can play music on hold to the caller. Julian. On 6/9/05, James Bean [EMAIL PROTECTED] wrote: Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. ___ 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] Group/Broadcast Voicemail
Here's what it looks like Robert -- Executing VoiceMail(SIP/6153245827-0a2e, [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]@mcdstores) in new stack -- Playing 'vm-intro' (language 'en') -- SIP/6153245805-d694 answered SIP/207.65.117.4-bf434468 -- Attempting native bridge of SIP/207.65.117.4-bf434468 and SIP/6153245805-d694 -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: wav49, 0x958fc80 -- x=1, open writing: /var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: gsm, 0x9590c48 -- x=2, open writing: /var/spool/asterisk/voicemail/mcdhq/801/INBOX/msg format: wav, 0x94e4358 -- User hung up Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 9 09:27:51 NOTICE[21651]: app_voicemail.c:1242 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] You can see there's about 33 voicemail accounts but it will only copy to about 22 of the boxes. Robert Goodyear wrote: On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote: So, anyone else have any ideas? I tried the below suggestion and it's still only sending out 20 of the 32 voicemails. C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson [EMAIL PROTECTED] wrote: Still only doing 20 voicemails. Thanks for the suggestion. - Here's a weird idea. Can you put each group of 20 users into a distribution group whose distributOR is a member of a distribution group itself? Pseudo-diagram, assuming: 400 is the master VM broadcaster and 5600 through 5631 are your 32 users. exten = 400,1,VoiceMail(u401402403) exten = 401,1,VoiceMail(u560056015602...5619) exten = 402,1,VoiceMail(u562056215622...5639) Wonder if that would work? Robert Goodyear Brand Up LLC http://www.brand-up.com
RE: [Asterisk-Users] Lingo(.com) and Asterisk
According to the fab sheet for the Dlink router they provide, it's SIP with G711, G723, G726, G729. Order the service, get the router, plug the WAN port into your LAN, fire up Ethereal, power up the router, and sniff what's being passed, you might be able to determine the user/pass, IP and codec and then you can just proxy Asterisk in with the same user/pass and bob's your uncle. hth -Original Message- From: Bas Rijniersce [mailto:[EMAIL PROTECTED] Sent: Thursday, June 09, 2005 8:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Lingo(.com) and Asterisk Hello, A long Google search didn't turn any clear answer. Does somebody use Asterisk in combination with Lingo? Thank you, Bas ___ 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
Re: [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch
Thanks again Julian. Quick update. Worked great with 1.07 (Which is good since cvs head gave me hell today) On Thu, 2005-06-09 at 08:55 +0100, Julian J. M. wrote: I've made a backport of this patch for asterisk stable. You can get it here: http://www.maxosystem.net/asterisk . The page is in Spanish, but you just need to download and apply the patch to chan_zap.c. It also works with bristuff patch applied. Julian J. M. On 6/9/05, Neil and Fiona [EMAIL PROTECTED] wrote: Is there a list of options that are valid for stable? I downgraded from Head to stable when I had IAX trunking problems (one way audio) with a VSP. So I am using my conf files from Head, which could be the problem. I've got a copy of sample config files from 1.07 (Or I think they are, I didn't label it well when I archived it). It seems to have the option in it. There has been a patch in Head for the IAX2 trunking problem, so I think I could go back to head. ___ 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] Incoming call stops at random with Teliax
Rich and anybody else on Teliax might want to check a couple of times. I
have seen a few people having the same issue in the last couple of weeks.
We have been seeing this if we do random tests between 5-60 min.
I have tried one other thing in combination with Rich's config is to use one
of the IP address(208.139.204.232)instead of the FQDN which has two
different address.
So far this seems to be working.
Teliax form
http://www.teliax.com/forum/viewtopic.php?p=438#438
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Thursday, June 09, 2005 5:59 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using for the last several months:
[teliax]; for incoming calls
context=teliax-incoming
type=user
auth=md5
secret=mymd5secret
disallow=all
allow=gsm
trunk=no
[teliaxout] ; for outgoing calls
type=peer
host=voip.teliax.com
username=myname
auth=md5
secret=mymd5secret ; provided by teliax
disallow=all
allow=gsm
trunk=no
Calls are then placed using something like:
; Calls directed to Teliax.com
; long distance calls completed via Teliax.com
exten = _1NX,1,SetCallerID(3035551212|a)
exten = _1NX,2,SetCIDName(MyName|a)
exten = _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
exten = _1NX,4,Congestion
If I recall correctly, the majority of the above was provided in an
email from teliax when I signed up.
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[Asterisk-Users] having to reload asterix after internet connection failure
Hi, I've been having some problems with my internet connection, it cuts out for aprox 30 seconds at a time and after that i have to do a reload in asterix for it to re-register my sip account with broadvoice otherwise it won't accept any connection till i reload, is there a way for it to automatically re-register or am I missing something else ? Thx. Armand ___ 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] E1 and SS7
The telephone company in Honduras say they will only supply an E1 circuit with SS7 signaling. Has anyone else run into this? Can anyone recommend a work-around for this problem? Thanks ___ 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] DID on SIP channel
Hi Olle. Exactly! Im using Nortel SIP server to register an * box. Could you point me on some documentation about this new feature? Thanks a lot. Denis. On 08 de jun de 2005, at 07:52, Olle E. Johansson wrote: I guess you are registering with the Nortel SIP server? All the incoming calls will go to the incoming extension you are registering with them. If they add aliases for several incoming lines to one registration, you need to check the To: header. This is only possible in CVS head with the SIP get header function in the dial plan. This is one of the reasons I am planning to implement a type=service object in sip.conf ___ 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] Cisco 7960 and Skinny
Title: Message Hi, I have bought two Cisco 7960 phones. I have tried to set-up them to work with Asterisk over Skinny protocol, but when I try to dial the phone from Asterisk it says that all lines are busy. Is there something that should be configured on the phone's side? Can someone help me with that? Also,I would like to upgrade these phones to use SIP. How can I get the SIP firmware for my phones. I have tried at Cisco web site but I couldn't find firmware downloads. Can someone help me with that? The phones are currently using following firmware: Application Load ID: P003AM30 Boot Load ID: PC030300 Regards, Stojan ___ 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] Cisco 7960 and Skinny
I have bought two Cisco 7960 phones. I have tried to set-up them to work with Asterisk over Skinny protocol, but when I try to dial the phone from Asterisk it says that all lines are busy. Is there something that should be configured on the phone's side? Can someone help me with that? try chan_sccp cvs -d:pserver:[EMAIL PROTECTED]:/cvsroot/chan-sccp login cvs -z3 -d:pserver:[EMAIL PROTECTED]:/cvsroot/chan-sccp co -P chan_sccp cd chan_sccp vi Makefile and edit the asterisk path make and make install Also, I would like to upgrade these phones to use SIP. How can I get the SIP firmware for my phones. I have tried at Cisco web site but I couldn't find firmware downloads. Can someone help me with that? You need to pay for a service contract to get the sip firmware from the cisco site Sergio ___ 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] Re: AgentCallBacklogin (logout continued...)
1 2 [EMAIL PROTECTED] wrote:
Anyone know if
- it is possible to limit 1 agent per extension where
the last agent to log in overrides any previous agents
or
- a Command/application to clear all agents logged in
on extension
Does this look like it would require a custom mod to
do it?
In Asterisk v1, the biggest stumbling block to implementing this in the
dial plan is the fact that logging out of an extension requires you to
enter your agent password. This is really silly, and an unnecessary
security risk (imagine if you left yourself logged into a unix shell,
and it was impossible to log out without a password- it's begging for
others to use your account instead of being a good samaritan and
logging you out).
I implemented a solution to the problem you describe, using only the
dial plan. This is possible, though a bit awkward, only if you don't use
agent passwords. Entering an extension of # (no extension) in the
AgentCallbackLogin application logs you out.
The basic solution I implemented is: before logging an agent in, check
to see if any other agent is logged in to the same extension. If they
are, log that agent out instead of logging the new one in.
Unfortunately, logging out an agent automatically hangs up, so the agent
needs to call back again if they want to log back in.
I'm not fully happy with this solution! I would prefer a better one if
any exists, but I haven't found anything better suited to my needs on
the wiki or elsewhere yet, which works for Asterisk v1.
This is only deployed in our test environment, and hasn't been stress
tested with actual people yet. (And now that I look back on it, there
are some obvious optimizations I could make which were apparently not
obvious at the time...)
Here is a macro I set up to log an agent out of whatever extension
they're logged into. The agent is an agent channel name, not an
extension number. The dial # automatically hack came from the wiki;
just specifying an agent extension of # doesn't work.
[macro-agent_logout]
; Agent logout.
;
; Log out the specified agent.
;
; On a logged-in agent phone, caller ID is set to the agent's caller
; ID automatically, so we don't need to look up the agent ID for this
; caller ID.
exten = s, 1, setglobalvar(agent=${ARG1})
exten = s, 2, noop(agent ${agent})
exten = s, 3, dial(local/[EMAIL PROTECTED]/n,,D(w#))
exten = logout, 1, noop(agent logout ${agent})
exten = logout, 2, wait(1)
exten = logout, 3, agentcallbacklogin(${agent},@shared_phones)
This macro logs an agent off of a specified extension. Note that the
AGENTBYCALLERID variable is only accurate if exactly ONE agent logs into
an extension. If more than one logs in, this will only log out the last
agent who logged in. This makes the later agent login hack necessary...
(macro-answer_wait just does an answer() and wait(1).)
[macro-agent_logout_ext]
exten = s, 1, setvar(agent=${AGENTBYCALLERID_${ARG1}})
exten = s, 2, gotoif(${agent}?3:101)
exten = s, 3, macro(agent_logout,${agent})
exten = s, 101, macro(answer_wait)
exten = s, 102, playback(agent-loggedoff)
exten = s, 103, hangup()
For agent logins, I use this macro.
[macro-agent_login]
; Agent login.
; If someone is already logged in to this extension, then turn this
; into an agent logout. Otherwise, log in: we only prompt for agent
; ID, and we don't use passwords.
;
; ${ARG1} is the full caller ID of the extension the agent will be
; logged in to.
;
; ${ARG2} is the CALLERIDNUM of the extension the agent will be logged
; in to.
; If there's an agent set for this callerid, then log it out;
; otherwise, log in.
exten = s, 1, setvar(agent=${AGENTBYCALLERID_${ARG1}})
exten = s, 2, gotoif(${agent}]?104:9)
exten = s, 9, noop(logging in ${ARG2})
exten = s, 10, agentcallbacklogin(,[EMAIL PROTECTED])
; Agent is logged in, log them out. Unfortunately we can't then log back
; in because it hangs up.
exten = s, 104, goto(agent_logged_in,s,1)
[agent_logged_in]
; An agent is already logged in. Press 1 to log out, or any other
; button to cancel.
exten = s, 1, macro(answer_wait)
exten = s, 2, background(agent_logged_in)
; this hangs up when it's finished.
exten = 1, 1, macro(agent_logout,${agent})
exten = _[2-9#*], 1, playback(goodbye)
exten = _[2-9#*], 2, hangup
And finally, the actual agent service extensions:
exten = 212, 1, macro(agent_login,${CALLERID},${CALLERIDNUM})
exten = 213, 1, macro(agent_logout_ext,${CALLERID})
I hope this helps. Please feel free to forward any questions you may have.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
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RE: [Asterisk-Users] More than one account from the same provider?
Try this... get rid of the friend and use the peer and user defs. Then pay close attention to which parameters apply to those two defs. Change the [teliax] to something different, like [teliax-in] and [teliax-out]. All that will do is separate the configs for incoming and outgoing, I need two completely separate incoming and outgoing accounts. Yes, that's true, but I thought you already stated that you have two accounts, right? The above comments went beyond the two account issue and was attempting to suggest that a number of implementors have had an understanding problem with friends, peers, and users. From my perspective (having gone through some of the same understanding problems), splitting the stuff into peer and user has been very very good in promoting the understanding part. On top of that, there are definite differences in how friend, peer, and user is implemented (or functions) between sip and iax. So given your original verbage, if you have two accounts, split the friend (for both) into peer and user (for both) and define the codec you want for both accounts. ___ 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] REPOSTED: Polycom 500 Group Call Pickup Feature and *
If you activate (via sip.cfg) the feature Group Call Pickup, its no surprise that asterisk doesn't know what to do with this feature request. But it is being sent as a SIP SUBSCRIBE request, and I'm wondering if, as asterisk stands, there is a way to take advantage of this feature to emulate the *8# normal behavior. If anyone has any input, there is also a call parking function that I think is SIP SUBSCRIBE-based. Here is the 'sip debug' snippet from when I pressed the New Call - Pickup - Group softkeys: Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Max-Forwards: 70 Expires: 0 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=5041eff0 Content-Length: 0 to 192.168.0.234:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms morse*CLI Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Proxy-Authorization: Digest username=201, realm=asterisk, nonce=5041eff0, uri=sip:[EMAIL PROTECTED]:5060, response=b48b989d85958a6ce18c9431058ce6f3, algorithm=MD5 Max-Forwards: 70 Expires: 0 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Looking for groupcallpickup in default Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.234:5060 Destroying call '[EMAIL PROTECTED]' morse*CLI sip no debug SIP Debugging Disabled Chris Coulthurst [EMAIL PROTECTED] ___ 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
[Asterisk-Users] howto write CDRs on two mysql servers
For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario ___ 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] DISA Help
The problem was the dtmf, changed to rfc2833 and it works like a beauty. setup the galaxyvoice for the incoming(free) and route calls through sunrocket and broadvoice. thanks, bill On 6/8/05, Wilson Pickett [EMAIL PROTECTED] wrote: when i try to dial a number it just dies.Meaning what? Silence? Hangup?Does dialing voicemail on that same setup work? That would tell whether it hears the DTMF.Other wise, check the codec and dtmf mode, some combinations don'twork on some phones.___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
Re: [Asterisk-Users] CallerID/chan_sccp
Joseph wrote:
On Thu, 2005-06-09 at 02:24 +1000, Julien Goodwin wrote:
On 8/06/2005 11:37 PM, Sergio Chersovani wrote:
Joseph ha scritto:
When sending a call to a line defined on chan_sccp, there is an
error on the console that says:
Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
Fixed, you can find the patch here
http://www.c-net.it/chan_sccp/
And this has been committed, should work through in about 5 hours
(thanks sourceforge)
It works.
Thanks.
I just downloaded the latest chan_sccp and am having problems with
internal to internal calls with callerid. I added a few debug lines to
the code to help sort it out, but here's what happens... Exten 581
calls 580. On the display 581 shows Unknown number to 580. On exten
580, the display shows Test Phone2 to Unknown number. Here are some
of the lines from the CLI including my added debug lines:
-- Set calledParty Name: Test Phone1 Number 580
-- Executing Dial(SCCP/581-0005, SCCP/580|15|Ttr) in new stack
SCCP trying to call SCCP, format 4, data, 580
-- --* 581
-- New channel context: office
-- Asterisk request to call: SCCP/580-0006
-- Set callingParty Name: Test Phone2 Number 581
== Sending Packet Type SetLampMessage (16 bytes)
== Sending Packet Type SetRingerMessage (8 bytes)
== {CallStateMessage} callState=RingIn(4), lineInstance=1, callReference=6
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name: Test Phone2
*** Calling Party Number: 581
*** Called Party Name:
*** Called Party Number:
== Sending Packet Type CallInfoMessage (208 bytes)
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {SelectSoftKeysMessage} lineInstance=1 callReference=6
softKeySetIndex=3 validKeyMask=65535/65535
== Sending Packet Type SelectSoftKeysMessage (20 bytes)
-- Called 580
-- Asked to indicate '3' (Dialing) condition on channel
SCCP/581-0005
-- Current tone (36) is equiv to wanted tone (36). Ignoring.
== Sending Packet Type DisplayPromptStatusMessage (48 bytes)
== {CallStateMessage} callState=RingOut(3), lineInstance=1,
callReference=5
== Sending Packet Type CallStateMessage (28 bytes)
*** Calling Party Name:
*** Calling Party Number:
*** Called Party Name: Test Phone1
*** Called Party Number: 580
The lines beginning with *** are the debug lines I added inside the
sccp_channel_send_callinfo function. Any ideas?
Mark
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Re: [Asterisk-Users] having to reload asterix after internet connection failure
I've been having some problems with my internet connection, it cuts out for aprox 30 seconds at a time and after that i have to do a reload in asterix for it to re-register my sip account with broadvoice otherwise it won't accept any connection till i reload, is there a way for it to automatically re-register or am I missing something else ? Pure guess, if you're using a nat box at your asterisk end, add the qualify statement to see if that helps. ___ 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] Polycom IP-500 600 Nat settings.
I have looked at the wiki and the mailing list. But I need to find how do we setup the external IP address and the rtp ports for the Polycom IP-500 and IP-600. There web interface has a nat setting but can't find instructions on how to set this up. I would like to set this up via there ftp file setup instead of via there web setting. Also There QoS settings are set to 5 and 2 but there it does not say if you change it to 7 or to a lower number which one gives you better priority. Main problem I am having is that the polycoms work great as long as there on the same LAN. once they go through a Nat router even if all the ports are open we get one way audio or no audio. The asterisk servers are on a real world IP address and the Phones are behind a Nat firewall called m0n0wall. We have all ports open going out to where the asterisk box is setup. ___ 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] Group/Broadcast Voicemail
On Jun 9, 2005, at 7:33 AM, Chris Stinson wrote: Here's what it looks like Robert -- Executing VoiceMail(SIP/6153245827-0a2e, [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]8 [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]83 [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]840 @mcdstores[EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]845@ mcdstores[EMAIL PROTECTED]@mcdstores) in new stack -- Playing 'vm-intro' (language 'en') -- SIP/6153245805-d694 answered SIP/207.65.117.4-bf434468 Do you think there's any coincidence that exten 838, where you indicate the last vm is copied to, falls right around character 256 of that argument? I would experiment by temporarily shortening the contexts to q (for headquarters) and s (for stores) and trying again. That would shorten the argument you're sending to the vm app considerably and would give proof if this is or isn't the issue. Let me know... I'm very curious now! Robert Goodyear Brand Up LLC http://www.brand-up.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
Re: [Asterisk-Users] howto write CDRs on two mysql servers
Rosario Pingaro wrote: For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario The quickest way would be to make a copy of cdr_addon_mysql and rename the app and conf file, recompile, then load that module. You would basically be running 2 instances of that module. Long term solution would be to rewrite cdr_addon_mysql to support multiple databases. -Matthew ___ 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] astGUIclient installation problem
Hi everyone: I try to install astGUIclient for my call center. I'm interesting to put in work the monitoring client, i follow step by step the installation from scratch but when i try to run the application from my Windows XP astGUIclient i got the follow error: Client does not support authentication protocol requested by server; consider up grading MySQL client at astGUIclient_1.1.0.pl line 4704 Any idea will be appreciated. Regards. Kritikus. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] having to reload asterix after internet connection failure
a simple workaround is to put a cronjob to execute #!/bin/bash asterisk -rx 'reload' of course, i think that the best choice is find out why is needed to reload, i dont think that is a normal behaviour best regards On 6/9/05, Armand Sulter [EMAIL PROTECTED] wrote: Hi, I've been having some problems with my internet connection, it cuts out for aprox 30 seconds at a time and after that i have to do a reload in asterix for it to re-register my sip account with broadvoice otherwise it won't accept any connection till i reload, is there a way for it to automatically re-register or am I missing something else ? Thx. Armand ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] CallerID/chan_sccp
t downloaded the latest chan_sccp and am having problems with internal to internal calls with callerid. I added a few debug lines to the code to help sort it out, but here's what happens... Exten 581 calls 580. On the display 581 shows Unknown number to 580. On exten 580, the display shows Test Phone2 to Unknown number. Here are some of the lines from the CLI including my added debug lines: What type of cisco phone is this? It does use both Calling and Called info on the display. I'll patch it to fill both on outgoing and incoming call. I'll fix it later Sergio ___ 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] More than one account from the same provider?
On Jun 9, 2005, at 6:47 AM, Chris Mason (Lists) wrote:
Try this... get rid of the friend and use the peer and
user defs.
Then pay close attention to which parameters apply to those two defs.
Change the [teliax] to something different, like [teliax-in]
and [teliax-out].
All that will do is separate the configs for incoming and outgoing, I
need
two completely separate incoming and outgoing accounts.
Chris: you've answered your own question then. You'd have to convince
Teliax to send a different authentication name to your server. That's
why I was trying to clarify whether you meant outbound or inbound.
Given that we're talking inbound, I feel you're stuck.
Teliax could theoretically allow users to have a specific auth name
(could be as simple as [TELIAX-{username}] ) that their switch DIALs
against, but we're delving into territory where six of us on the planet
would want this and couldn't even come close to ever making it cost
effective for them to make such a change to their code.
Right?
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Definately problems with voice quality and caller ID is not working very well. I have e-mail a couple times and still no response from their tech support on this. This is very concerning since I tried all 3 servers with the same results. On 6/8/05, Julio Arruda [EMAIL PROTECTED] wrote: Roman Zhovtulya wrote: Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? I've been using VOIPJET for Brazil LD without any problems. (or should I say, my wife has been using, still can't thank VOIP enough for the savings..) ___ 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
RE: [Asterisk-Users] Incoming call stops at random with Teliax
Top posting to keep with the flow...
I'd have to guess at a couple of things here. Its already been stated that
asterisk _only_ uses the first of multiple dns A records when queried.
It would appear the voip.teliax.com dns A records point to 208.139.204.232
and 208.139.204.228, so one would have to guess they are attempting to
load balance their servers based on dns. But, asterisk will use the
first one returned for the duration of asterisk's uptime.
However, what happens if you register with one of those two servers and
the second server attempts to _complete_ a call to your asterisk box?
If you're behind a nat box, the answer is likely to be dependent on
how that box is set up. If you're not behind a nat box, one still
might have an issue if using the type=friend and iax (eg, does iax
find a matching context based on ip, username, last context in iax.conf,
etc).
I'd have to guess that by using a specific IP address, you've managed
to find a work-around for that problem, but there is likely an
underlying root-cause that has yet to be identified.
Rich
Rich and anybody else on Teliax might want to check a couple of times. I
have seen a few people having the same issue in the last couple of weeks.
We have been seeing this if we do random tests between 5-60 min.
I have tried one other thing in combination with Rich's config is to use one
of the IP address(208.139.204.232)instead of the FQDN which has two
different address.
So far this seems to be working.
Teliax form
http://www.teliax.com/forum/viewtopic.php?p=438#438
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Thursday, June 09, 2005 5:59 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using for the last several months:
[teliax] ; for incoming calls
context=teliax-incoming
type=user
auth=md5
secret=mymd5secret
disallow=all
allow=gsm
trunk=no
[teliaxout] ; for outgoing calls
type=peer
host=voip.teliax.com
username=myname
auth=md5
secret=mymd5secret; provided by teliax
disallow=all
allow=gsm
trunk=no
Calls are then placed using something like:
; Calls directed to Teliax.com
; long distance calls completed via Teliax.com
exten = _1NX,1,SetCallerID(3035551212|a)
exten = _1NX,2,SetCIDName(MyName|a)
exten = _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
exten = _1NX,4,Congestion
If I recall correctly, the majority of the above was provided in an
email from teliax when I signed up.
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Re: [Asterisk-Users] E1 and SS7
I have the exact same problem.It would ideal if we could set an astersik box with 2 E1 ports to do an IP-to-SS7 conversion. Anyone has done this before? C. Savinovich At 11:08 AM 6/9/2005, you wrote: The telephone company in Honduras say they will only supply an E1 circuit with SS7 signaling. Has anyone else run into this? Can anyone recommend a work-around for this problem? Thanks ___ 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
RE: [Asterisk-Users] Asterisk Live! CF
Seshu, Are you working on a VIA based motherboard? I am working on a VIA based motherboard. Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite good when emulating i686 behavoir and since his distro is compiled for i686... We are trying to confirm that but may be interesting to know about your setup and how is Kristian's distro compiled. On Mon, 6 Jun 2005 16:41:43 -0400, Kanuri, Seshu (Company IT) wrote Kristian, I am talking about your distro, that does not seem to be able to boot when I have mounted (if that is the right word) the CF into my Dell Server and tried to boot from it as the only IDE drive available. The Linux just does not kick in. If you want to debug this I can Fedex to you, my 800MB CF disk with your distro on it, you for your RD. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Monday, June 06, 2005 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Live! CF abel wrote: My theory is that the 64 MB image is built with a specific hdd form factor and when burning onto a different size CF it is mapped differently and it does not work. On the other hand, you always can find out how the device is beeing seen by the system and customize the binary image accordingly. Other software prepared to be run from CF (I recall WISP, the LEAF branch for wireless routers) have a final step which takes the software already compiled and 'packages' it to build the disk image. I would be extremely happy if I could download the code tree and run that final step by myself to get the disk image that suits my needs. Second best would be to get the source tree and compile all the stuff to get that point. Is that possible? Is the code available in the way I need for this operation? TIA. abel, This is simply untrue. My distro's (AstLinux) 32mb CF images work on anything... http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3 -- Kristian Kielhofner NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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
Re: [Asterisk-Users] having to reload asterix after internet connection failure
On Thu, 2005-06-09 at 11:20 -0500, Moises Silva wrote: a simple workaround is to put a cronjob to execute #!/bin/bash asterisk -rx 'reload' in /etc/ppp/ip-up.local put service asterisk reload each time the connection is made then * will reload. -- Dave Cotton [EMAIL PROTECTED] ___ 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] Anyone noticed Voipjet voice quality problems?
Any other confirmation for this problem? My service seems to be fine but I have not completed a long duration call yet. I had a user complain last week about call degradation after 5-10 minutes but that has been it. I will test some more and let you know. I am on the west coast server. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Thursday, June 09, 2005 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? Definately problems with voice quality and caller ID is not working very well. I have e-mail a couple times and still no response from their tech support on this. This is very concerning since I tried all 3 servers with the same results. On 6/8/05, Julio Arruda [EMAIL PROTECTED] wrote: Roman Zhovtulya wrote: Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? I've been using VOIPJET for Brazil LD without any problems. (or should I say, my wife has been using, still can't thank VOIP enough for the savings..) ___ 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 ___ 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] Notice Message
You don't have a context called 'default'. Several parts of asterisk will default to going to that context unless specified. Usually it will be empty for security reasons do to that. Not certain what part from the error message is trying to reach it, but creating a empty default context will hopefully give you a more clearer warning/error/notice message. --johann craz sead wrote: hi all i have a notice message that comming frequently says that pbc.c:1329 pbx_extention_helper; cannot find extention context 'default' anyone know this warning and how to solve because its realy anoying thks roy __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ 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] Agent refuses to log out
Well, sort of :) We have agents using the AgentLoginCallBack functionality. The agents log in using their agent number, with the extension automatically entered for them. When they log out, they again use the AgentLoginCallBack app, but using just a # for the new extension (logs them out). Occasionally, an Agent simply refuses to log out. You get a message That Agent Is Already Logged On A Agent Logoff Agent/xxx from the CLI does not work either. Looking at the channels, I see pbx*CLI show channels Channel (ContextExtensionPri ) State Appl. Data Local/[EMAIL PROTECTED],1 (AgentQ s1 )Down (None) (None) 1 active channel(s) I am making an assumption when I say that this can't be right - I thought that there would always be 2 channels at least (source and dest). What is causing this ? How can I hang this channel up ? Is this channel causing the Agent logoff problem ? - Oh, the Agent cannot logon either :( Julian ___ 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] Asterisk Live! CF
abel wrote: Seshu, Are you working on a VIA based motherboard? I am working on a VIA based motherboard. Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite good when emulating i686 behavoir and since his distro is compiled for i686... We are trying to confirm that but may be interesting to know about your setup and how is Kristian's distro compiled. i586-MMX and higher. The Soekris Net4801 is an i586, and the mini-itx's are not good with i686 instructions... -- Kristian Kielhofner ___ 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] Clicks in audio with TE100P PRI
On Thu, Jun 09, 2005 at 12:51:30AM -0300, Alejandro G wrote: I should tell you that the TE100P is connected to another E1 board (not a live E1) from Natural Microsystems which acts as a gateway to PSTN. This board works as a PRI master but I don't think that this could be the problem as long as using other phones or in LAN it works perfectly and the voice is clear with no clicks o sound looses. Do you find that these clicks occur at the same time concurrently with increased hard drive activity? If so, and if you have an IDE hardrive, try doing a `hdparm -u1 /dev/yourhardrivedevice` Matthew Fredrickson ___ 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] astGUIclient installation problem
Hello, This issue was just handled Monday on the astguiclient-users list: http://sourceforge.net/mailarchive/forum.php?thread_id=7448401forum_id=4358 6 You just need to use OLD_PASSWORD in the SET PASSWORD for your mysql server to get the auth method for that account back to the pre 4.1.12 version default method of login authentication. Also, consider joining the astguiclient-users list, a lot of tweeks and fixes come up on there that don't make it into the documentation right away. MATT--- -Original Message- From: kritikus Araklidas [mailto:[EMAIL PROTECTED] Sent: Thursday, June 09, 2005 12:11 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] astGUIclient installation problem Hi everyone: I try to install astGUIclient for my call center. I'm interesting to put in work the monitoring client, i follow step by step the installation from scratch but when i try to run the application from my Windows XP astGUIclient i got the follow error: Client does not support authentication protocol requested by server; consider up grading MySQL client at astGUIclient_1.1.0.pl line 4704 Any idea will be appreciated. Regards. Kritikus. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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
Re: [Asterisk-Users] Asterisk Live! CF
On Thursday 09 Jun 2005 17:29, abel wrote: Seshu, Are you working on a VIA based motherboard? I am working on a VIA based motherboard. Andy Powell (author of Asterisk Live! distro) tells me that VIA is not quite good when emulating i686 behavoir and since his distro is compiled for i686... We are trying to confirm that but may be interesting to know about your setup and how is Kristian's distro compiled. The Via processors emulate the i686 just fine. The problem has always been with GCC. B ___ 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] having to reload asterix after internet connection failure
actually this is a freebsd box and its not behind NAT, i'm not sure if it has to do with my internet connection anymore. Right now it says : bsd*CLI sip show registry HostUsername Refresh State sip.broadvoice.com:5060 username 3032 Registered However when I call it, it doesnt seem to hit the box at all, it keeps ringing and then broadvoice says nobody is available to take the call. Now if i do a reload and call it will work temporarly. Any suggestions ? -Armand On 6/9/05, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2005-06-09 at 11:20 -0500, Moises Silva wrote: a simple workaround is to put a cronjob to execute #!/bin/bash asterisk -rx 'reload' in /etc/ppp/ip-up.local put service asterisk reload each time the connection is made then * will reload. -- Dave Cotton [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] howto write CDRs on two mysql servers
why not just use mysql replication to the second one? On 6/9/05, Rosario Pingaro [EMAIL PROTECTED] wrote: For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario ___ 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
Re: [Asterisk-Users] Asterisk to Cisco Unity
Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ 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
[Asterisk-Users] Voicemail and MS Exchange Synchronization
We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this being developed for Asterisk and yet it would appear to be a critical component needed to migrate customers used to fully integrated Unified Messaging systems to Asterisk. (a) Has anyone cracked this nut (or started on it)? (b) Anyone interested if we post a bounty? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ 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] howto write CDRs on two mysql servers
On Jun 9, 2005, at 8:51 AM, Rosario Pingaro wrote: For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Might be easier to add a second host as a replica server with the mySQL Administrator. Might lessen the load on Asterisk by not waiting on a second, remote connection. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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
[Asterisk-Users] RE: astGUIclient installation problem
Hi everyone: I try to install astGUIclient for my call center. I'm interesting to put in work the monitoring client, i follow step by step the installation from scratch but when i try to run the application from my Windows XP astGUIclient i got the follow error: Client does not support authentication protocol requested by server; consider up grading MySQL client at astGUIclient_1.1.0.pl line 4704 At the risk of being a jerk, did you try to find the answer on your own? http://www.google.com/search?hl=enq=Client+does+not+support+authenticat ion+protocol+requested+by+server%3B+btnG=Google+Search I just copied the first bit of the error message into a Google search box. Lots of information. This error usually means you are running a 4.1+ version of MySQL server, and the client doesn't understand the newer authentication protocol. You need to set the password using the OLD_PASSWORD function in MySQL. Take a look at the top entry when you run the Google search, as it is directly from MySQL's manual. This should fix the error. Good luck. And in the future, you can save time by trying a really quick Google search on error messages. ___ 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] Lingo(.com) and Asterisk
fire up Ethereal, power up the router, and sniff what's being passed, you might be able to determine the user/pass, IP and codec Proxy and codec is simple enough. But the authentication is via a challenge-response on SIP so that will be a lot harder, if you have the computing power to crack it. --Luki ___ 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] Voicemail and MS Exchange Synchronization
-Original Message-
From: George Pajari [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 09, 2005 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization
{clip}
(a) Has anyone cracked this nut (or started on it)?
I'm not aware of any proof of concept code, but I understand the discussion to
be in the direction of changing app_voicemail to use a maildir directory
format, thereby allowing the voicemail to be easily exposed and accessed by a
third party unix mail server program that is unaware of Asterisk (eg. Cyrus
IMAP).
Actually getting it synced directly into your Exchange server backend seems far
more challenging. We had been eagerly looking forward to an IMAP access method,
hence being able to add an IMAP account to Outlook on our workstations and
accessing voicemail that way (some people use the vm-as-attachment with the
Delete option at the moment, others just get new vm notices).
If you're looking at a large deployment client side integration would quickly
become a maintenance nightmare... Do you have any pointers as to how the
backend Exchange integration process actually works?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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