[asterisk-users] SPA400 and asterisk
Hi Guys I am just looking to see if you can help me. I have been investigating the SPA400 and it seems to run asterisk for the voicemail system. Does anyone know if it could be programmed to also talk to the FXO ports? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SPA400 and asterisk
Yes you can but I am looking to see if asterisk can use the FXO ports directly. The proprietary SIP to FXO gateway on there is not as good as if you could use asterisk with the FXO ports directly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo (M) Sent: Tuesday, 12 June 2007 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SPA400 and asterisk You could use it as a usually FXO Gateway. I have tested and it works fine. 2007/6/12, MBIT Technologies [EMAIL PROTECTED]: Hi Guys I am just looking to see if you can help me. I have been investigating the SPA400 and it seems to run asterisk for the voicemail system. Does anyone know if it could be programmed to also talk to the FXO ports? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Small CDR Billing Program
Hi Guys I am just looking around for a small billing program but can't really find what I am looking for. It needs to bill straight off the CDR. It should grab all the CDR records from the asteriskcdrdb mysql database then have a rates table to that it calculate a bill from. Is there any open source packages or commercial packages that will account for billing say only 5 extensions? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Trixbox vs. Custom install
If you are installing in a business environment I think being able to use both and know the benefits of both are pretty essential. FreePBX is a great tool and should be used to its potential because it has some great features. It can also lessen the time it takes to do an install. Some have said that FreePBX is a control freak and wants control of everything. This is not true. If you know how to manipulate freepbx properly then you shouldn't have any trouble adding in custom features as well as using the main features built into FreePBX. Trixbox can have some problems with hardware because it is a prebuilt RPM. If you can't install asterisk and other components from source you really shouldn't be doing production installs. Being able to patch the source with different features is also very essential. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 13 February 2007 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Re: Trixbox vs. Custom install IMHO, If you don't know enough about Linux/Asterisk/FreePBX to be able to set it up yourself you should not be doing it for a Production install in a business environment. NOTE: Production install in a business environment does NOT include setting it up in your house with extensions for the kids and wife! That is what Trixbox is for. -Original Message- From: Edward Halman [mailto:[EMAIL PROTECTED] Sent: Monday, February 12, 2007 12:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Trixbox vs. Custom install Hi Stefano, I am a proponent of the step-by-step installation on a complete linux distribution. Like someone said in another posting, the GUIs are nice, but isolate you from the .conf files to the point where customization can be a bit tricky. However, Trixbox w/ FreePBX and A2Billing works out of the box with very little patching or configuration needed. If A2Billing is all he/she anticipates needing to do, FreePBX is a mature and stable and he/she will probably be happy with it. FreePBX is a bit of a chore to install and configure without Trixbox, if you don't have a solid understanding of dependencies, linux security, apache and MySQL. Same with A2Billing. Edward Halman (718) 705-7451 [EMAIL PROTECTED] -- Message: 12 Date: Mon, 12 Feb 2007 17:42:15 +0100 From: Stefano Corsi [EMAIL PROTECTED] Subject: [asterisk-users] Trixbox vs. Custom install To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 45CB1BDE00381960@ (added by [EMAIL PROTECTED]) Content-Type: text/plain; charset=us-ascii; format=flowed Hello, I'm following the thread [EMAIL PROTECTED] vs Trixbox, and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom step by step installation on a distribution of his/her choice? Thanks Stefano -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 with 1 FXO
Hi David Also make sure the power connector is also connected to the board. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Friday, 9 February 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] TDM400 with 1 FXO Hi, Yes it should, I have changed it back and is still causing the same problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Hi All, I cannot get my TDM to work correctly. In my /etc/zaptel.conf file I have loadzone = us defaultzone=us fxoks=1 Shouldn't this be fxsks if you're using an FXO module as analog trunk? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI scripts
Ive tried to install a couple of applications like the teleyapper script on nerdvittles but I seem to be getting an error executing the AGI script. I have also tried installing the voiceone gui and have problems executing the AGI scripts in this. Here is the error == Spawn extension (yapper-gen, s, 5) exited non-zero on 'IAX2/250-2' -- Executing DeadAGI(IAX2/250-2, teleyapper.php|0|1|1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/teleyapper.php -- AGI Script teleyapper.php completed, returning 0 The question Im wondering is, is there any common problems with executing AGI? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI Dropouts (Solved)
Hi Kevin Yes you shouldn't need the busy detect in there for digital systems. Busydetect is only really useful on analog systems which can't detect incoming hangup. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Tuesday, 15 August 2006 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Dropouts (Solved) After much playing and getting nowhere, I was on the phone to the guys from www.voipshop.com.au and mentioned that the pri dropout problem was occuring and if they had any solutions. Immediately they mentioned something that causes a problem in australia. On longdistance phone calls (sometimes) you hear a series of short beeps that indicates to the receiver that it's a long distance call. Few calls seem to do this these days but some do. It seems to be exactly what the busy detect code looks for. On removing the busydetect (or rather setting it to 0) it solved our dropout problems. Just thought I would let everyone know incase it's an issue for anyone else. Regards Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ever donate Software to Digium? If you did youra fool.
If you gave software to Digium then you helped Mark become very rich. What's wrong with making Mark a rich man? He has come up with a great new product and I'm sure he has risked a lot to get it to you. Asterisk is free so he owes you nothing. How about you take your jealousy elsewhere or maybe put your energy into doing something worthwhile. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Gardiner Sent: Wednesday, 9 August 2006 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ever donate Software to Digium? If you did youra fool. With respect, I don't think you understand the dynamics of growing a business. If we are all to benefit from the continued development of Asterisk then it is in our own best interests for Digium to succeed, because their success is for our benefit. Your posting is unfortunate as it disregards the considerable effort, cost and time put into Asterisk by Mark and Digium. By the way, I have no relation with Digium other than to derive a considerable benefit from open source software developed by Mark/Digium and a lot of other programmers, for which I am extremely grateful. Digium is not being given a whole load of money - the investors will want a slice of the company and the future profits. That's how VC funding works. Randall H. wrote: If you gave software to Digium then you helped Mark become very rich. http://abcnews.go.com/Technology/wireStory?id=2290152 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Uni Call
Hi Guy Does anyone know where I can find a patch for the latest unicall and asterisk 1.2.7.1 Regards Mark Brooker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall libmfcr
Hi Im wondering if you can help me with this error mfcr2.c:3543: error: `UC_REQUESTMOREINFO_ORIGINATING_NUMBER' undeclared (first use in this function) mfcr2.c: In function `call_control': mfcr2.c:3894: error: `UC_OP_REQUESTMOREINFO' undeclared (first use in this function) mfcr2.c:3895: error: `uc_requestmoreinfo_t' undeclared (first use in this function) mfcr2.c:3895: error: syntax error before ')' token make[1]: *** [mfcr2.lo] Error 1 make[1]: Leaving directory `/usr/src/unicall/libmfcr2-0.0.3' make: *** [all] Error 2 Ive got the latest snapshot 20060205. Regards Mark Brooker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall in Australia
Hi Has anyone set this up and what protocol variant did you use for unicall? There is no Australian setting that I know of so any help would be greatly appreciated. Regards Mark Brooker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and NEC NEAX 2000 IPS
Hi Guys I am just looking for a bit of help here. I am trying to integrate the 2 of these together via a E1 link. The link has no signalling and is basically a dumb 2 meg link. Does anyone have any zaptel configurations they could give me. I am using a Digium TE110P. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Any Polycom dealers willing help out?
Hi Heres the details of a ftp server that I have put them on Server: ftp.voipeye.com.au User: [EMAIL PROTECTED] Pass: update Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 024950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: Thursday, 6 July 2006 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Any Polycom dealers willing help out? Hi All, We are in search of the latest Polycom firmware SIP 1.6.6 as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only older versions. Are there any kind Polycom authorized dealers who can help me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Asteirsk-Users]TE110P configuration problem
If you are plugging the card into a PSTN line then I think signalling needs to be pri_cpe Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of root linux Sent: Thursday, 6 July 2006 3:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [Asteirsk-Users]TE110P configuration problem Here is my E1 setting in zaptel.conf: - span=1,1,0,ccs,hdb3 bchan=1-15 # set this to 1-15,17-31 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 loadzone = us defaultzone=us Here is my E1 setting in zapata.conf: - switchtype=euroisdn signalling=pri_net group=1 context=default channel = 1-15,17-31 --- Maximo Villamayor [EMAIL PROTECTED] wrote: Hi list, I have a TE110P card with asterisk at home 2.5 I can't make to work it with my Telco provider, but i have a router cisco 2600 series with a E1 interface working correctly with the same provider. The configuration of router with the E1 card is: ! controller E1 1/1 framing NO-CRC4 ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-semi-compelled ani cas-custom 0 country paraguay use-defaults ! How it would have to be my configuration file (zaptel.conf) for my TE110P card? Thanks for the help! =) Max Villamayor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ISDN BRI NetJet
For a single BRI the Netjet is the way to go. I have the card running with the mISDN drivers. I haven't tried capi yet but seems to work great in ptp mode. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: Sunday, 18 June 2006 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: ISDN BRI NetJet On Sat, Jun 17, 2006 at 02:33:26PM -0300, Hermann Wecke wrote: I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 It could work with the deprecated chan_modem. Don't wast your time. Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? Replace it by some Cologne Chip Card. The single port card is cheap. Than you can use bristuff, chan_misdn or visdn. Assuming you can get such a card where you live. In Australia there is no PCI HFC adapter available. We can get the AVM fritz card, but the importers have marked the price up quite substantially based on the fact that they have A-Tick approval on it. But as per my last post, the netjet mISDN drvier doesn't currently work, so if you can find an alternate card where you are then do that. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, mISDN and a Fritz card
Have you followed the wiki entry on aussievoip? You are better to use capi because I don't think misdn actually supports fax. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Jones Sent: Friday, 9 June 2006 6:35 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk, mISDN and a Fritz card Hi, I've been having problems getting Asterisk to work with a fritz card (in ptp mode). We've managed to get everything configured, however, as soon as the call is connected we get Unhandled message: prim 281 len -22 from addr 100, dinfo 0 on this port I checked, and mISDN_dsp is loaded. Any help would be much appreciated! Regards, Chris Jones // Network Administrator Top Level Internet e: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, mISDN and a Fritz card -- kernel crashes
What kernel are you using Chris? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Jones Sent: Friday, 9 June 2006 10:36 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk,mISDN and a Fritz card -- kernel crashes Who mentioned fax? :-) Native CAPI isn't an option for us (unfortunately), as our service is a point-to-point service, which the AVM CAPI drivers don't support. We've now got another problem -- we're now able to make calls -- once. The kernel panics as soon as someone terminates a call (and, so it would seem, at various other times too). This has only been occuring since mISDN was installed. Any ideas? Thanks, Chris Message: 4 Date: Fri, 9 Jun 2006 18:39:36 +1000 From: MBIT Technologies [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk, mISDN and a Fritz card To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Have you followed the wiki entry on aussievoip? You are better to use capi because I don't think misdn actually supports fax. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SIP 1.6.6
Send me your details and I can give you a ftp to download it from. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 024950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob McKrill Sent: Wednesday, 7 June 2006 3:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SIP 1.6.6 I'd suggest calling whoever you buy your phones from. The distributor I work with requires that you are Polycom certified to be able to purchase phones from them, but once you are certified with Polycom you can actually download the firmware from their extranet. On 6/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially? Not that Polycom is analy retentive, or anything like that... Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 or asterisk
Can MAD crash a server like mpg123 can? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Tuesday, 30 May 2006 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] mpg123 or asterisk Can't you use mpg123 as compiled under x86_32? I do on a few servers I have. I found madplay better process wise than mpg123. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: 29 May 2006 21:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 or asterisk Well, being unable to compile mpg123 under x86_64 i installed lame and transformed the mp3--wav--raw. and using files as the format player. Are there any good scripts to stress test MoH? I want to test this machine for 1000 calls on hold. http://www.asteriskguru.com/tutorials/astertest.html AsterTest is good but i dont have access to another * installation. On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: Please let us know your results. I cannot really test this in production system since it is a $16,000/hr call center. I was using madplay but it was crashing and creating zombie processes, I figured native was not the way to go since all of the different audio streams. Mpg123 works perfectly for me under a load of sixty channels, I can confirm that for sure. Thanks, Steve Erick Perez wrote: Interesting. So, i will have to test then... On 5/27/06, Steve Totaro [EMAIL PROTECTED] wrote: In my very limited testing of native, each channel was receiving a different stream (each caller heard something different). Under a high volume of calls, which is going to hurt performance more? Transcoding MP3s but sending a single stream or separate streams per call under native? When I say high, I mean 1,000+ calls. Thanks, Steve Erick Perez wrote: Thanks to all. Native format will be. On 5/27/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: Vahan Yerkanian wrote: Erick Perez wrote: should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. mpg123 was used back when asterisk didn't have native format support. If you are expecting heavy load, the native format is the way to go. You might decide not to use mp3 format at all, recompressing your MoH files using sox to the formats you gonna use, such as .al, .ul, .gsm, or leave it at .sln to cut the decoding leg only. Heh, damn this GPRS connection. In order to pass the time while downloading messages I reply before they are all in, and yet by the time I have received all the messages I note that your question has already been answered! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mISDN FAX
Thanks for that Ill look into it. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ralf Schlatterbeck Sent: Friday, 19 May 2006 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mISDN FAX On Wed, May 17, 2006 at 02:29:12PM +1000, MBIT Technologies wrote: -- Executing Goto(mISDN/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(mISDN/1-1, FROM_DID=s) in new stack -- Executing Set(mISDN/1-1, FAX_RX=system) in new stack -- Executing Set(mISDN/1-1, [EMAIL PROTECTED]) in new stack -- Executing Answer(mISDN/1-1, ) in new stack -- Executing PlayTones(mISDN/1-1, ring) in new stack -- Executing NVFaxDetect(mISDN/1-1, 4) in new stack -- Executing Goto(mISDN/1-1, ) in new stack == Spawn extension (ext-did, s, 7) exited non-zero on 'mISDN/1-1' Hmm, maybe you should leave out the PlayTones above?? I'd guess the remote fax still thinks the line is ringing and is not making any noises that would allow fax detection. Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN FAX
Hi Guys I have mISDN installed and working correctly but I am unable to receive a fax through the connection. Is there anything I need to do to get this running? I have nvfaxdetect installed but I think this only works with sip, iax and zaptel Any help is greatly appreciated. Regards Mark Brooker ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mISDN FAX
Hi Avi I can't see any fax detection at all in my call logs. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Wednesday, 17 May 2006 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mISDN FAX On 17/05/2006, at 1:26 PM, MBIT Technologies wrote: I have mISDN installed and working correctly but I am unable to receive a fax through the connection. I have NVFaxDetect and RxFAX running on my CAPI channels, so I know that works. :) Does NVFaxDetect detect the fax correct? Does it drop to the fax = exten? cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mISDN FAX
This is what Im getting when I try to receive a fax -- Executing Goto(mISDN/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(mISDN/1-1, FROM_DID=s) in new stack -- Executing Set(mISDN/1-1, FAX_RX=system) in new stack -- Executing Set(mISDN/1-1, [EMAIL PROTECTED]) in new stack -- Executing Answer(mISDN/1-1, ) in new stack -- Executing PlayTones(mISDN/1-1, ring) in new stack -- Executing NVFaxDetect(mISDN/1-1, 4) in new stack -- Executing Goto(mISDN/1-1, ) in new stack == Spawn extension (ext-did, s, 7) exited non-zero on 'mISDN/1-1' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Wednesday, 17 May 2006 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mISDN FAX On 17/05/2006, at 1:45 PM, MBIT Technologies wrote: I can't see any fax detection at all in my call logs. What does your dialplan look like for incoming calls? Do you give NVFaxDetect enough time? I find that 4 seconds is good, but 2 seconds is dodgy, for example. cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mISDN FAX
By the way im running FreePBX 2.1beta1. There could be some issues so I may need to upgrade. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Wednesday, 17 May 2006 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mISDN FAX On 17/05/2006, at 1:45 PM, MBIT Technologies wrote: I can't see any fax detection at all in my call logs. What does your dialplan look like for incoming calls? Do you give NVFaxDetect enough time? I find that 4 seconds is good, but 2 seconds is dodgy, for example. cYa, Avi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on amd SERVER
You shouldn't really have any problems with i386 version on the AMD. When centos moved to 4.3 the x86_64 bit version was a mess with trying to install packages. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne Sent: Friday, 5 May 2006 4:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on amd SERVER Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreePBX in production?
Basically you can change the ones which don't have _additional.conf at the end. So with your extensions.conf you can change the extensions.conf and extensions_custom.conf Best to keep all your configs in the custom file though. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, 1 May 2006 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX in production? So, how do you know which conf files one can hand edit versus those that might be overwritten? You may only change the *_custom.conf files. :) And the *_additional.conf files are the ones overwritten by the config in the DB. So you can edit the other ones. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I have one of the Draytek USB adaptors. Can someone point me in the right direction on how to get mISDN running with it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, 17 March 2006 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Any idea where I can get some of these units in Melbourne? Paul Hales AsteriskIT Faxing received by SpanDSP seems to work fine with these units. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Tuesday, 14 March 2006 9:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe HI Craig and all that is following this. I am running a Vanilla 2.6.11 From cli, misdn show config Misdn General-Config: - VERSION: 0.2.1 - DEBUG_LEVEL: 1 - TRACEFILE: not set - TRACE_CALLS: false - TRACE_DIR: /var/log/ - BRIDGING: no- STOP_TONE_AFTER_FIRST_DIGIT: yes - APPEND_DIGITS2EXTEN: yes- L1_INFO_OK: yes - CLEAR_L3: no- DYNAMIC_CRYPT: no - CRYPT_PREFIX: **- CRYPT_KEYS: test,muh So Far, no dropped calls etc Todays testing will be faxing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, 13 March 2006 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) Both of them operating in ptmp with no echo cancel turned on at this ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
I'm not sure if this has been brought up before but I don't think Asterisk is the package you would want to use if you are going to implement a fax production server. A more mature package in this area is Hylafax. I think you will find it will be far more reliable than Asterisk in the case of a fax production server. Heres a link http://www.hylafax.org Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Scardine Sent: Wednesday, 5 April 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk in production as a fax server,anyone? Don Pobanz escreveu: Frame slips are NOT motherboard related! I had problems with some combinations of motherboards, memory sizes and linux kernel versions. There are timing problems that also causes frame slips, like buffer overruns or underruns, but these are software related. -- Paulo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501 Output volume
Hi Guys Is there anyway to adjust the output volume on the Polycom 501? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED] and H323
Hi The H323 patch for [EMAIL PROTECTED] is very out dated. Try http://www.mbit.com.au/h323/h323.zip It should have everything you need to get H323 up and running. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Wednesday, 8 March 2006 1:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and H323 On Tue, 2006-03-07 at 12:08 +0200, Viktor Tatianin wrote: Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': If you have asterisk 1.2.4 version you must have to compile oh323 as in http://www.oinko.net/astrecipes/index.php?n=40 but replacing the versions from: http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib- Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh3 23-Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh 323-0.7.3.tar.gz Please help for resolve this problem Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 501
I guess it doesnt work by default on my phone. You still need to press hash to transfer calls. The transfer button doesnt work. Where do I set it? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, 2 March 2006 3:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom 501 Those buyttons do work with asterisk by default... what kind of problems are you having? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MBIT Technologies Sent: Wednesday, March 01, 2006 7:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom 501 Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 501
AMP is being run but it seems the transfer needs to be configured in the phone somewhere so when you press the transfer button its like hitting #. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Aufflick Sent: Thursday, 2 March 2006 10:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 One thing to keep in mind when someone says Asterisk does that by default is that a lot of people have AMP installed, and an AMP installation includes extra configuration and features as well as the web interface. It may be that there is phone-specific config installed with AMP that is not installed in a base Asterisk installation. Cheers, Mark. -- Mark Aufflick e: [EMAIL PROTECTED] w: www.pumptheory.com (business) w: mark.aufflick.com (personal) p: +61 438 700 647 f: +61 2 9436 4737 On 3/2/06, MBIT Technologies [EMAIL PROTECTED] wrote: I guess it doesn't work by default on my phone. You still need to press hash to transfer calls. The transfer button doesn't work. Where do I set it? Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Anton Krall Sent: Thursday, 2 March 2006 3:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom 501 Those buyttons do work with asterisk by default... what kind of problems are you having? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MBIT Technologies Sent: Wednesday, March 01, 2006 7:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom 501 Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. iBurst Wireless Broadband from $34.95/month - Platform Networks Spam Virus Filtering by Mail Security To report SPAM forward the spam message to: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 501
Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk compile error
Hi Guys I have a problem compiling Asterisk 1.2.4. I am getting this error make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Has anyone come across this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp
Hi Guys Im just trying to patch asterisk with the apps_makefile.patch to get fax working. I am getting this error when trying to patch [EMAIL PROTECTED] apps]# patch Makefile.patch patching file Makefile Hunk #1 FAILED at 48. Hunk #2 FAILED at 77. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej Does anyone have a working patch? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Capi problem
Hi Guys Im having a problem getting CAPI to work on my Traverse NetJet card. CAPI is enabled in the kernel and Im using the mISDN drivers with the NetJet patch. I cant seem to get astcapi to load Heres the output im getting Nov 12 21:18:36 VERBOSE[4011]: == Registered application 'WaitMusicOnHold' Nov 12 21:18:36 VERBOSE[4011]: == Registered application 'SetMusicOnHold' Nov 12 21:18:36 VERBOSE[4011]: [chan_capi.so]Nov 12 21:18:36 VERBOSE[4011]: [chan_capi.so] = (Common ISDN API for Asterisk) Nov 12 21:18:36 VERBOSE[4011]: == Parsing '/etc/asterisk/capi.conf': Nov 12 21:18:36 VERBOSE[4011]: == Parsing '/etc/asterisk/capi.conf': Found Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64) Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64) Nov 12 21:18:36 WARNING[4011]: CAPI not installed, CAPI disabled! Nov 12 21:18:36 WARNING[4011]: chan_capi.so: load_module failed, returning -1 Nov 12 21:18:36 VERBOSE[4011]: == Unregistered channel type 'CAPI' Nov 12 21:18:36 WARNING[4011]: Loading module chan_capi.so failed! I think it could be a udev problem so I put a file called 10-capi.rules in my udev directory with the following SYSFS(dev)=68:0, NAME=capi20 SYSFS(dev)=191:[0-9]*, NAME=capi/%n When I do a capiinit it says ERROR: cannot load module kernelcapi Does there need to be another entry for kernelcapi? Any help would be greatly appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
