Re: [Asterisk-Users] SIP Phones-Receptionist Setup
$400-500 device here. Not very price competitive. I would like to see less than half that. What is the price point you are trying to hit? Any piece of a proprietary telecom system is by nature overpriced to begin with, and receptionist consoles certainly fit into that category. I agree that any touch screen ought to be able to do normal computer graphics. At this point, you are into normal LCD displays with touch capability, which I know retail over US$500 even for smaller ones. And at that point, you are back to doing a double-display on the receptionist computer, and in reality, you could directly run something like that FOP that everyone seems on about (if it fits your needs), since as I understand (never having programmed them myself) that touching a spot on a screen is the same as clicking a mouse there in terms of window-manager events. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Sun, Nov 21, 2004 at 12:21:16AM -0700, Kevin P. Fleming spake thusly: > And there are tons of extremely small systems that could do this job. I > have here in front of me a Soekris net4801 which is tiny, noiseless I know there are plenty of small systems that would be great. The problem is the cost of the display really. Interfacing the display via USB could be a real problem. I would really prefer something with VGA so I could do real graphics on the display. And there are plenty such boards also. The Soekris net4801 costs $197 in units of 100 on soekris.com for the bare board. Then you would need a case, display, handset...We're looking at a $400-500 device here. Not very price competitive. I would like to see less than half that. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpeq5BIKxBGy.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice
> are they really /unlimited/ in the truest sense of the word ? > US$24.95, even if it's only for unlimited calls to Malaysia > (where i am) seems very, very attractive. when something is this > attractive, i start looking for the catch. AFAIK, no one offers truly unlimited service. Companies differ greatly in openness of the contractual details. I subscribe to "unlimited" (POTS domestic US) long distance from SBC. The contract clearly states that monthly usage exceeding 5000 minutes is billed at $0.04 per minute. Not cheap, but it won't break you if go a little over. At the other extreme, there are many horror stories of Vonage customers whose service was terminated, without warning, for excessive usage. Broadvoice appears to be somewhere in between. I am considering their service, and called them to ask about allowed usage. They would not disclose their limits, but when I mentioned that my calls typically run 2500-3000 minutes per month, mostly to the US, they said that this was well below their "alarm" levels. There may be a technical problem with Broadvoice for your application. I suspect that all calls proxy the media stream through their server (in the US). Perhaps a Broadvoice customer can confirm or deny this. If that's the case, the roundtrip delay on your calls to Malaysia will include *four* hops across the Pacific (~400 milliseconds). If there's any echo, it will be very disconcerting. Even if not, you'll have problems when both parties start talking at about the same time. You can use their free trial offer to see if the delay is bothersome. --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
Tracy R Reed wrote: This is the way I want to go. A very small PC with a good touch screen. And there are tons of extremely small systems that could do this job. I have here in front of me a Soekris net4801 which is tiny, noiseless computer (similar to a PC, but not quite the same) that draws nearly no power. It has a USB port, so if the display/touchscreen can be interfaced via USB you'd be all set. There are also less expensive versions of this unit )with slower CPUs, but still fast enough for this application), down to around $125 or so including case and power supply. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Sun, Nov 21, 2004 at 06:18:04PM +1300, Matt Riddell spake thusly: > Me and another guy are working on LCD drivers etc for Linux. The thing > is, the display would be run from your Asterisk Server. I.E. It will > need to be run from either Parallel, Serial or USB port. We will open What would this do? Would it be touch screen or just display status info (busy/available etc) on phone lines? This would mean the receptionist would have to have the PBX under her desk which is a bad idea. I always insist it go in a datacenter or at least an out of the way phone closet where it won't get kicked or messed with. > day would do it...the problem is finding a spare day. I guess the other > option would be to use one of the small PC's to run Asterisk and a panel > on the receptionists desk. This is the way I want to go. A very small PC with a good touch screen. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpQqN2JelBKt.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Sun, Nov 21, 2004 at 12:05:27AM -0500, Gregory Junker spake thusly: > What is the size of the current line panel on her desk? I am thinking it > might be worthwhile to produce an addon to Asterisk that drives a flat > touchpanel that does the same thing as the current solution. Baby steps. I have indeed considered this. They have around 15 extensions. I just don't have the time to develop this device at the moment and if I did I'm not sure how professionally I could package it or how cheap I could make it. Small touch screen displays seem pretty expensive and I haven't looked into whether X supports touchscreens. A small and cheap Via Epia board connected to the network with one of these displays sitting right on top of it would be an awesome device though. In fact I would love to turn something like this into a general phone solution by adding a handset/speakerphone, and putting a soft phone on it (thus making it a hard phone since it is a dedicated hardware device since all voip phones are just software running on a hardware platform anyhow) and sell it as the ultimate flexible voip phone. The only problem is that it would cost way too much unless one built it in volume which isn't likely. Might look kinda hokey too if you just put a touch screen on top of a box and attach a handset and speakers to it. I don't know how people would react if it weren't a curvy looking slick pastic thing. I would really like to see a general purpose phone platform running Linux with a nice touchscreen where I could customize the interface and everything though. Their phone system is still not complete because of this issue. Turned out the Cisco 7914 I originally ordered to do this does not work with the SIP image so I had to send it back and get a Snom 220 which I will use with the hint extenstion etc as described here: http://www.voip-info.org/wiki-Asterisk+phone+SNOM http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html and hope it works like I expect it to. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpt8LCAOehN6.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
Me and another guy are working on LCD drivers etc for Linux. The thing Including touchscreen? Ideally someone would tell me how to make something either a) seamlessly convert serial/parallel/USB port to TCP and back at the other end, or b) point me to a resource on a simple chip with TCP support that will maybe print out 8-bit packets to an 8-bit pin out. Ideas? http://www.digi.com/products/terminalservers/index.jsp Works terribly well in my experience. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
Gregory Junker wrote: Most customers don't want to be in a new era. They want something they are accustomed to. I don't need any more impediments to making money than I've already got. So if the customer wants a busy lamp, I am going to do my best to give it to them. I agree. This is why engineers do not make good salesmen. What is the size of the current line panel on her desk? I am thinking it might be worthwhile to produce an addon to Asterisk that drives a flat touchpanel that does the same thing as the current solution. Baby steps. If she can use the current mechanical switchboard then she can use this with no real retraining...plus you get the additional benefit of flexibility in configuration (if they end up needing more lines than the current panel supports, this is just a software change). Greg Me and another guy are working on LCD drivers etc for Linux. The thing is, the display would be run from your Asterisk Server. I.E. It will need to be run from either Parallel, Serial or USB port. We will open source it once finished, and are not too far off, probably just a spare day would do it...the problem is finding a spare day. I guess the other option would be to use one of the small PC's to run Asterisk and a panel on the receptionists desk. Ideally someone would tell me how to make something either a) seamlessly convert serial/parallel/USB port to TCP and back at the other end, or b) point me to a resource on a simple chip with TCP support that will maybe print out 8-bit packets to an 8-bit pin out. Ideas? Also, someone did an interface for ShoutBox or something similar, if it was designed well, you may be able to just plug in a line status module. (you'd have to write it of course!). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice
BroadVoice is the most competitive companies I've seen on the net for residential users, it's why I chose them for my own personal home service. I mean, $19.95 a month, unlimited USA and 21 countries. $24.95 a month, unlimited USA and 35 countries. are they really /unlimited/ in the truest sense of the word ? US$24.95, even if it's only for unlimited calls to Malaysia (where i am) seems very, very attractive. when something is this attractive, i start looking for the catch. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Queue Sounds - not working?
Andy Rosen wrote: I just did a chmod 777 for all the queue sound files in /var/lib/asterisk/sounds, and have the same error. OK, then permissions aren't the issue. I read the error as to say that it couldn't find the file as opposed to your interpretation that it was found, but not accessible. Any other suggestions? Not immediately coming to mind, no. Can you Playback() that same file in a simple extension for testing? app_queue uses the same functions for playing sounds as Playback(), so it would be good to know if it fails under both cases. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
Most customers don't want to be in a new era. They want something they are accustomed to. I don't need any more impediments to making money than I've already got. So if the customer wants a busy lamp, I am going to do my best to give it to them. I agree. This is why engineers do not make good salesmen. What is the size of the current line panel on her desk? I am thinking it might be worthwhile to produce an addon to Asterisk that drives a flat touchpanel that does the same thing as the current solution. Baby steps. If she can use the current mechanical switchboard then she can use this with no real retraining...plus you get the additional benefit of flexibility in configuration (if they end up needing more lines than the current panel supports, this is just a software change). Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Sat, Nov 20, 2004 at 09:25:38PM -0600, Brian Roy spake thusly: > I would look at putting a dual monitor on her desk. You can pick up a > 15" flat panel and a video card for about the same cost as the SNOM. She doesn't want another monitor. > Asterisk is not your dad's pbx. Most customers don't want to be in a new era. They want something they are accustomed to. I don't need any more impediments to making money than I've already got. So if the customer wants a busy lamp, I am going to do my best to give it to them. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpGzQsPPVj0G.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and scansoft TTS
Hi list, Looking for a good TTS with french voice, only scansoft's realspeak seems to fill the bill (though at a high price), but I'm having a hard time trying to find technical details. More than one year ago there was a discussion about * and high quality speech synthesis, and people indeed mentionned scansoft products. So I'd like to know if people actually have working installations. To be more specific, I'd like to use it for for an IVR using AGI/EAGI and Perl scripts; is it doable, what is the impact in terms of processing power, memory needed, linux distro... for say a 10 simultaneous channels. Do I need to buy their very expensive Linux SDK, or just licences? Thanks in advance for sharing your experience. Jean-Denis ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 dead but pid file exists - gdb asteriskcore.13089
UPDATE UPDATE UPDATE... 9/22 Also make clean. bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of nor amie aris > Sent: Saturday, November 20, 2004 10:13 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Asterisk dead but pid file exists - gdb > asteriskcore.13089 > > Dear ALL, > > Any clues or tips for the following gdb messages. > > [EMAIL PROTECTED] asterisk]# uname -a > Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct > 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux > > localhost*CLI> show version > Asterisk CVS-HEAD-09/22/04-11:19:09 built by > [EMAIL PROTECTED] on a i686 running Linux > > [EMAIL PROTECTED] asterisk]# gdb asterisk core.13089 > GNU gdb Red Hat Linux (5.3.90-0.20030710.41rh) > Copyright 2003 Free Software Foundation, Inc. > GDB is free software, covered by the GNU General > Public License, and you are > welcome to change it and/or distribute copies of it > under certain conditions. > Type "show copying" to see the conditions. > There is absolutely no warranty for GDB. Type "show > warranty" for details. > This GDB was configured as > "i386-redhat-linux-gnu"...Using host libthread_db > library "/lib/tls/libthread_db.so.1". > > > > Core was generated by `asterisk -vvvg -c'. > Program terminated with signal 11, Segmentation fault. > Reading symbols from /lib/libdl.so.2...done. > Loaded symbols for /lib/libdl.so.2 > Reading symbols from /lib/i686/libpthread.so.0...done. > Loaded symbols for /lib/i686/libpthread.so.0 > Reading symbols from /usr/lib/libncurses.so.5...done. > Loaded symbols for /usr/lib/libncurses.so.5 > Reading symbols from /lib/i686/libm.so.6...done. > Loaded symbols for /lib/i686/libm.so.6 > Reading symbols from /lib/libresolv.so.2...done. > Loaded symbols for /lib/libresolv.so.2 > Reading symbols from /lib/libssl.so.4...done. > Loaded symbols for /lib/libssl.so.4 > Reading symbols from /lib/i686/libc.so.6...done. > Loaded symbols for /lib/i686/libc.so.6 > Reading symbols from /lib/ld-linux.so.2...done. > Loaded symbols for /lib/ld-linux.so.2 > Reading symbols from /usr/lib/libgpm.so.1...done. > Loaded symbols for /usr/lib/libgpm.so.1 > Reading symbols from > /usr/lib/libgssapi_krb5.so.2...done. > Loaded symbols for /usr/lib/libgssapi_krb5.so.2 > Reading symbols from /usr/lib/libkrb5.so.3...done. > Loaded symbols for /usr/lib/libkrb5.so.3 > Reading symbols from /lib/libcom_err.so.2...done. > Loaded symbols for /lib/libcom_err.so.2 > Reading symbols from /usr/lib/libk5crypto.so.3...done. > Loaded symbols for /usr/lib/libk5crypto.so.3 > Reading symbols from /lib/libcrypto.so.4...done. > Loaded symbols for /lib/libcrypto.so.4 > Reading symbols from /usr/lib/libz.so.1...done. > Loaded symbols for /usr/lib/libz.so.1 > Reading symbols from > /usr/lib/asterisk/modules/chan_modem.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_modem.so > Reading symbols from > /usr/lib/asterisk/modules/chan_modem_aopen.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_modem_aopen.so > Reading symbols from > /usr/lib/asterisk/modules/res_musiconhold.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_musiconhold.so > Reading symbols from > /usr/lib/asterisk/modules/res_adsi.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_adsi.so > Reading symbols from > /usr/lib/asterisk/modules/res_features.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_features.so > Reading symbols from > /usr/lib/asterisk/modules/res_crypto.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_crypto.so > Reading symbols from > /usr/lib/asterisk/modules/res_indications.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_indications.so > Reading symbols from > /usr/lib/asterisk/modules/res_monitor.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_monitor.so > Reading symbols from > /usr/lib/asterisk/modules/res_agi.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/res_agi.so > Reading symbols from > /usr/lib/asterisk/modules/chan_sip.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_sip.so > Reading symbols from /lib/libnss_files.so.2...done. > Loaded symbols for /lib/libnss_files.so.2 > Reading symbols from /lib/libnss_nis.so.2...done. > Loaded symbols for /lib/libnss_nis.so.2 > Reading symbols from /lib/libnsl.so.1...done. > Loaded symbols for /lib/libnsl.so.1 > Reading symbols from /lib/libnss_dns.so.2...done. > Loaded symbols for /lib/libnss_dns.so.2 > Reading symbols from > /usr/lib/asterisk/modules/chan_modem_bestdata.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_modem_bestdata.so > Reading symbols from > /usr/lib/asterisk/modules/chan_modem_i4l.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_modem_i4l.so > Reading symbols from > /usr/lib/asterisk/modules/chan_agent.so...done. > Loaded symbols for > /usr/lib/asterisk/modules/chan_agent.so > Reading symbols from > /usr/li
[Asterisk-Users] Asterisk dead but pid file exists - gdb asterisk core.13089
Dear ALL, Any clues or tips for the following gdb messages. [EMAIL PROTECTED] asterisk]# uname -a Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux localhost*CLI> show version Asterisk CVS-HEAD-09/22/04-11:19:09 built by [EMAIL PROTECTED] on a i686 running Linux [EMAIL PROTECTED] asterisk]# gdb asterisk core.13089 GNU gdb Red Hat Linux (5.3.90-0.20030710.41rh) Copyright 2003 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-redhat-linux-gnu"...Using host libthread_db library "/lib/tls/libthread_db.so.1". Core was generated by `asterisk -vvvg -c'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/i686/libpthread.so.0...done. Loaded symbols for /lib/i686/libpthread.so.0 Reading symbols from /usr/lib/libncurses.so.5...done. Loaded symbols for /usr/lib/libncurses.so.5 Reading symbols from /lib/i686/libm.so.6...done. Loaded symbols for /lib/i686/libm.so.6 Reading symbols from /lib/libresolv.so.2...done. Loaded symbols for /lib/libresolv.so.2 Reading symbols from /lib/libssl.so.4...done. Loaded symbols for /lib/libssl.so.4 Reading symbols from /lib/i686/libc.so.6...done. Loaded symbols for /lib/i686/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/lib/libgpm.so.1...done. Loaded symbols for /usr/lib/libgpm.so.1 Reading symbols from /usr/lib/libgssapi_krb5.so.2...done. Loaded symbols for /usr/lib/libgssapi_krb5.so.2 Reading symbols from /usr/lib/libkrb5.so.3...done. Loaded symbols for /usr/lib/libkrb5.so.3 Reading symbols from /lib/libcom_err.so.2...done. Loaded symbols for /lib/libcom_err.so.2 Reading symbols from /usr/lib/libk5crypto.so.3...done. Loaded symbols for /usr/lib/libk5crypto.so.3 Reading symbols from /lib/libcrypto.so.4...done. Loaded symbols for /lib/libcrypto.so.4 Reading symbols from /usr/lib/libz.so.1...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /usr/lib/asterisk/modules/chan_modem.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_modem.so Reading symbols from /usr/lib/asterisk/modules/chan_modem_aopen.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_modem_aopen.so Reading symbols from /usr/lib/asterisk/modules/res_musiconhold.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_musiconhold.so Reading symbols from /usr/lib/asterisk/modules/res_adsi.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_adsi.so Reading symbols from /usr/lib/asterisk/modules/res_features.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_features.so Reading symbols from /usr/lib/asterisk/modules/res_crypto.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_crypto.so Reading symbols from /usr/lib/asterisk/modules/res_indications.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_indications.so Reading symbols from /usr/lib/asterisk/modules/res_monitor.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_monitor.so Reading symbols from /usr/lib/asterisk/modules/res_agi.so...done. Loaded symbols for /usr/lib/asterisk/modules/res_agi.so Reading symbols from /usr/lib/asterisk/modules/chan_sip.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_sip.so Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_nis.so.2...done. Loaded symbols for /lib/libnss_nis.so.2 Reading symbols from /lib/libnsl.so.1...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Reading symbols from /usr/lib/asterisk/modules/chan_modem_bestdata.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_modem_bestdata.so Reading symbols from /usr/lib/asterisk/modules/chan_modem_i4l.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_modem_i4l.so Reading symbols from /usr/lib/asterisk/modules/chan_agent.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_agent.so Reading symbols from /usr/lib/asterisk/modules/chan_mgcp.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_mgcp.so Reading symbols from /usr/lib/asterisk/modules/chan_iax2.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_iax2.so Reading symbols from /usr/lib/asterisk/modules/chan_local.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_local.so Reading symbols from /usr/lib/asterisk/modules/chan_skinny.so...done. Loaded sy
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Sat, 20 Nov 2004 15:58:48 -0800, Tracy R Reed <[EMAIL PROTECTED]> wrote: > I proposed something like this to a client but the receptionist has other > duties for her computer and does not want to have to have the operator > panel up all the time or go searching for the window in the taskbar every > time a call comes in. Nor does she want another computer on her desk > dedicated to just this. People don't like to change and to achieve maximum > success * needs to be able to replicate what the client is accustomed to. > I would look at putting a dual monitor on her desk. You can pick up a 15" flat panel and a video card for about the same cost as the SNOM. Not to mention, you get quite a bit more benifite from the FOP controls than you do busy lamp fields. It's a a new era here folks. Asterisk is not your dad's pbx. -Chuji ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Queue Sounds - not working?
I just did a chmod 777 for all the queue sound files in /var/lib/asterisk/sounds, and have the same error. I read the error as to say that it couldn't find the file as opposed to your interpretation that it was found, but not accessible. Any other suggestions? Andy - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, November 20, 2004 5:01 PM Subject: Re: [Asterisk-Users] Queue Sounds - not working? > Andy Rosen wrote: > > > Nov 20 16:40:32 WARNING[1120924864]: file.c:475 ast_openstream: File "queue-youarenext" does not exist in any format > > Nov 20 16:40:32 WARNING[1120924864]: file.c:779 ast_streamfile: Unable to open "queue-youarenext" (format GSM): No such file or directory > > > > Do queue sound files get pulled from a different location? > > No, they get pulled from the standard lib/sounds directory. > > Have you ensured that the user Asterisk is running as has access to > these files? It looks like the file is actually there, but Asterisk > could not open it (from the second message above). If the file had not > been there at all, this message would not have been generated. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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] ANY DEVELOPERS HERE?"warning:implicit declaration of function`__use_ast_pthread_create_instead__"
Jay Brussels said: > I did not intend to mix old with new. I ran "make clean". You still may be "mixing". I ran into the same a while back. The hitch for me was that the make files don't prefer to include from the source rather than from /usr/include. If you have a previous installation's files still in /usr/include/asterisk, they'll get used instead of the new ones in the source directory. I simply renamed (and removed later when deemed safe) /usr/include/asterisk before building the new version. Hope this helps, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions.conf help needed
1. I would like to setup my 4 extensions so that:
a. 601 rings first for 40 seconds
b. announcement that nobody picked up, and I will try another extension,
or caller may leave a message by pressing a key
c. 601, 602, 603 and 604 is ringing
d. after additional 40 seconds should come an announcement that nobody
picked up (hmm, what a surprise to tell them that), that the caller is
forwarded to the mailbox of 601
e. forward to mail box 601
alternate b and a swapped and say that the system trys now extension
601, but caller can leave a message (anytimem??? by pressing a key ??).
2. I have four DIDs from different countries and would re-write the
caller-ID so that I know from which line they are calling in.
E.g. if somebody calls from my Germany line, than I should see the phone
number 0049-{caller-id}, if somebody call me from UK, it should be
displayed as 0044-{caller-id}, if somebody call me from my toll free
number it should be displayed as 1-800-{caller-id}
Now it is even more tricky. Using ENUM it is possible, that somebody
call my 1-800 directly, and I would like to know that too. (It means if
I pay for the call or not)
The setup is now so, that if I dial 0044- it goes directly to the UK
phone line.
bye
Ronald
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Re: [Asterisk-Users] * and NAT
For anyone with the same issue , I just used IAX instead of SIP for external incoming connections to * . Cheers , John Khina . On 20/11/2004, at 10:34 PM, Wilson Pickett wrote: I'm having issues with * and getting connections outside of my local lan . My * server is sitting behind a firewall which performs NAT . In the general section if sip.conf srvlookup = yes nat = yes won't hurt. qualify sometimes causes problems, I'd remove it until all is working, but when used should take a number as in qualify=300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] ANY DEVELOPERS HERE? "warning:implicit declaration of function `__use_ast_pthread_create_instead__"
I did not intend to mix old with new. I ran "make clean". - Original Message - From: "Roger Schreiter" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, November 20, 2004 6:46 PM Subject: Re: [Asterisk-Users] ANY DEVELOPERS HERE? "warning:implicit declaration of function `__use_ast_pthread_create_instead__" Jay Brussels schrieb: > ... > When trying to upgrade to 1.0.2, I get several compile warnings such as: > chan_zap.c:3515: warning: implicit declaration of function > `__use_ast_pthread_create_instead__' Hi, I assume you mixed some parts of an older asterisk version with the new asterisk kernel. Did you recompile every module which should be loaded into the asterisk kernel? Maybe search for pthread_create in older source files and replace by ast_pthread_create! Check, whether the arguments remained the same ones and fix if necessary! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] res_config / realtime?
>From what I have been told, res_data stores *.conf files and not the actual data per se. RealTime is being written by the primary * developers while res_data is 3rd party. RealTime works seamlessly alongside *.conf files. Meaning you can have some configuration in a *.conf file and some in database and the app will work with both. I don't think res_data can do this. -Matthew - Original Message - From: "Tracy R Reed" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, November 20, 2004 4:52 AM Subject: Re: [Asterisk-Users] res_config / realtime? > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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] block caller id
> I have a PRI card. How do I block a caller id sent out to PSTN from a SIP
> client? I add a remote-party-id field "privacy=full" but still get caller id
> on a PSTN phone.
I have used the following successfully:
[dialout-local]
exten => _*67.,1,NoOp,${EXTEN}
exten => _*67.,2,CallingPres(35); 32 also works
exten => _*67.,3,Goto(dialout-local,${EXTEN:3},1)
( other dialout matches to send call to zap)
However with my telco, the PSTN user still sees the caller id
name, but the number is given as "private number"
bill
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Re: [Asterisk-Users] IAX issue at nufone
Hi, > Nufone provide me some config examples ... I can dialout > but I can't register my * Box, eg. whe I do "iax show registry" > I got only a "Request Sent" and later I have a "Timeout" First of all, I'm (and many are) sick to see blah blah blah doesn't work with blah blah blah. This is * ML, not nufone, not any other provider. then... can you ping switch-2.nufone.net ? Have you the corrent register statement into iax.conf? eg: [general] register => user:[EMAIL PROTECTED] I hope that user:passwd has been substituted with you account data, right? perhaps doing iax2 debug on * cli will help. or even send your config files matteo. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] system long multi argument
I've read other postings about "long arguments". Nothing is working for me. I'm trying to: exten => 5551212,1,Answer exten => 5551212,2,System(/usr/local/snpp.pl,-p Xpage -m some_text -r 8005551212) exten => 5551212,3,Hangup The script works from the shell. It also works from the extension if I leave off the arguments. This sends a page to an snpp server. Any ideas? I've tried multiple quotes, extra parens, writing to a text file and calling "snpp.sh". James Taylor MetroTel 903-793-1956 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to enter billing codes when dialling
AHBLWEB wrote: That's good and will get me the account#. I guess it's too much to ask for the same kind of tone prompts that occur after dialling the 77 (boop-boop) and then after the 7 digits (beep-beep-beep) before the user gets the final real dial tone? If you want another dial tone, set up some contexts and use DISA to give new dial tones. However, if you can retrain your users there are better ways to accomplish what you're after (i.e. just one DISA() with a password file) than emulating your old PBX exacatally. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly: > This does seem to be a common request, but I haven't seen any great Yes, it is. I am surprised * still can't do it. > Another option is the Flash Operator Panel, you can see a live demo at > http://www.asternic.com/ It is a flash applet that you can use in any I proposed something like this to a client but the receptionist has other duties for her computer and does not want to have to have the operator panel up all the time or go searching for the window in the taskbar every time a call comes in. Nor does she want another computer on her desk dedicated to just this. People don't like to change and to achieve maximum success * needs to be able to replicate what the client is accustomed to. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpTnWcELZh4L.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX IAX connection
Hi There I am trying to get the following setup going. Two PCs (gateways) running Linux FC2 both have aDSL connections running. And both run the Asterisk PBX software, I defined the IAX trunks in the iax.conf on both sides, the systems seem to call eachother but never get the handshaking completed ... The consoles keep reporting errors on both sides. Is there a example for doing this ?? -- A bartender is just a pharmacist with a limited inventory. I used to have a handle on life, but it broke --- Robert Bungener Dutch World Ventures ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel driver problem
i successfully compiled the zaptel drivers but when i try and load the module it fails gir:/usr/src/zaptel# modprobe zaptel gir:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_ec_chunk /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_unregister /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_alarm_notify /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_hooksig /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_transmit /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_receive /lib/modules/2.4.28/misc/wcfxo.o: /lib/modules/2.4.28/misc/wcfxo.o: unresolved symbol zt_register /lib/modules/2.4.28/misc/wcfxo.o: insmod /lib/modules/2.4.28/misc/wcfxo.o failed /lib/modules/2.4.28/misc/wcfxo.o: insmod wcfxo failed ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP Phones-Receptionist Setup
I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees? Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned key on the Snom 220¹s keypad, so if another call comes in they know he is on the phone instead of just blindly transferring the call and pushing the person to his voicemail. So they can ask the person hold or if them want to be transferred into Fred¹s voicemail. Any thoughts would be appreciated... I am new to the list, so my apologies if this has been addressed before. This does seem to be a common request, but I haven't seen any great answers yet. With the Snom phones, you can use the "hint() priority" (see http://www.voip-info.org/ and archives of this mailing list) for shared lines, but as far as I know, this only applies to sharing all the lines across all the extension. I don't know if it can apply to all the extra buttons that are on the Snom 220. Anybody tried that? Another option is the Flash Operator Panel, you can see a live demo at http://www.asternic.com/ It is a flash applet that you can use in any browser that lets you see all your lines and extensions and their current state. You can even do drag and drop call transferring. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ANY DEVELOPERS HERE? "warning: implicit declaration of function `__use_ast_pthread_create_instead__"
Jay Brussels schrieb: ... When trying to upgrade to 1.0.2, I get several compile warnings such as: chan_zap.c:3515: warning: implicit declaration of function `__use_ast_pthread_create_instead__' Hi, I assume you mixed some parts of an older asterisk version with the new asterisk kernel. Did you recompile every module which should be loaded into the asterisk kernel? Maybe search for pthread_create in older source files and replace by ast_pthread_create! Check, whether the arguments remained the same ones and fix if necessary! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing simple switch dialtone
If I choose immediate=no in zapata.conf, I can use the simple switch to get digits, but how can I change the dialtone played by the simple switch when I off hook? Leandro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd situation with Cisco 7960 IP phone
I have the cisco 7960 phone. If I call out on the phone I can hear the person called and they can hear me. If I have that same person and same phone call into my asterisk box and select my extension I answer as it is ringing then I cannot hear them speak. However, they can hear me. I dont understand? Anyone know what I am missing. I have 4 other sip phones on this machine (grandstream) and they work. I also have 3 IAXY's and they work. THanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with call files (/var/spool/asterisk/outgoing)
I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: .tiff|caller Note: All calls are going to the same fax machine, so some attempts on the second line will get a busy signal (there are two POTS lines in group 2). In the messages log, I get many of the following set: Unable to request channel Zap/g2/3036701917 Call failed to go through, reason 0 Can't change device with no technology! In message log I also get: Call completed to Zap/g2/3036701917 for the successful calls. Is there a workaround for this? If an outgoing call can't get a free line or gets a busy then I would want it to be retried later. I'm using today's cvs and spandsp-pre6. Thanks, Mike P.S. I've changed the call file to "Channel: Zap/27/3036701917" so it uses only one outgoing POTS line. With this change, I'm still getting the "Unable to request..." message set, but I'm not losing call files (the calls are being retried). All eight faxes are successfully delivered using one outgoing line. This tells me that getting a busy signal causes the call file to be removed and not retried. Bug? Cheers, -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TE410P PRI problems
Enoch Root wrote: However, even though I've configured extensions.conf to play a greeting message when dialed, I can't get it to work. When I type 'pri show channels' I see an output similar to this: PRI: Provisioned Line, Up, Active D-Channel: 16 With an extensive problem description like that I'm sure some of the more telepathic users of this list will be able to help. I, however, cannot. Your "pri show channels" output looks normal. What does "I can't get it to work" mean? Do you have _any_ output at all from a call attempt? What does zttool show for your span? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P PRI problems
Hi all! I'm trying to replace our old Siemens PBX with Asterisk. I've installed a TE410P card on a Linux box (Fedora Core 2 - 2.6.x kernel). I had no problem with compiling and installing modules and according to dmesg/zttool everything is ok. However, even though I've configured extensions.conf to play a greeting message when dialed, I can't get it to work. When I type 'pri show channels' I see an output similar to this: PRI: Provisioned Line, Up, Active D-Channel: 16 I wonder how an output of a correctly functioning system looks like. My zaptel and zapata config files looks like this: /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = pri_cpe group = 1 context = default channel => 1-15,17-31 __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Capi Deflection (CD) not working
Please, someone look into this. I really want CD to work. Thanks Jens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jens Hansen Sent: Tuesday, November 16, 2004 1:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Capi Deflection (CD) not working I did the following: - chan_capi-0.3.5/Makefile: uncommented CFLAGS+=-DDEFLECT_ON_CIRCUITBUSY - recompile asterisk + chan_capi - added /etc/asterisk/capi.conf: deflect=0800123456 ; some 0800 test number - in etc/asterisk/extensions.conf under [tcom-in]: exten => 98765,1,capiCD(0800123456) - made both b channels busy by outcalling on both lines (ISND BRI) - called my msn (98765) by mobile phone. => asterisk says: Nov 16 13:28:53 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:28:57 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:05 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:09 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nothing else. No deflection. No signs of doing anything. What else can i do? Thanks Jens Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 chan_capi-0.3.5 ISDN BRI / Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Queue Sounds - not working?
Andy Rosen wrote: Nov 20 16:40:32 WARNING[1120924864]: file.c:475 ast_openstream: File "queue-youarenext" does not exist in any format Nov 20 16:40:32 WARNING[1120924864]: file.c:779 ast_streamfile: Unable to open "queue-youarenext" (format GSM): No such file or directory Do queue sound files get pulled from a different location? No, they get pulled from the standard lib/sounds directory. Have you ensured that the user Asterisk is running as has access to these files? It looks like the file is actually there, but Asterisk could not open it (from the second message above). If the file had not been there at all, this message would not have been generated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] A new alternative to see who is online
It does seem to work with ZAP channels and releasing the source would be a great addition to Asterisk. I can see some User Interface improvments that could be made, but it appears to be a great foundation to work from as the basic functionality is there now. Lyle - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, November 20, 2004 4:45 PM Subject: Re: [Asterisk-Users] A new alternative to see who is online > Is there any chance that you might release the source code so that others can improve upon your code? I can see a real need for an application like this. I just wish it could be tweaked a little. > > -- > Jim Dossey Computer Services > > -- Original message -- > From: [EMAIL PROTECTED] > > Hi all, > > > > I have been facing about the problem to know who is online with asterisk PBX. > > However users wanted to see it right away, without launching any application. As > > I could not find any solution with IP phones and users were really complaining, > > I decided to write this little application that runs under windows and stays on > > screen. > > > > It is not perfect, but it works and I think it can help other people. > > > > Have a look: > > http://mapage.noos.fr/~b.nico/ > > > > Regards, > > Nicolas > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > 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 > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Sounds - not working?
Hello...I am just starting to play with queues in
Asterisk (CVS-v1-0-10/21/04-18:23:13) and have got everything working except for
sounds. I have defined the sounds (per the Wiki) in queues.conf as
follows:
queue-youarenext = "queue-youarenext" ; ("You are
now first in line.")queue-thereare = "queue-thereare" ; ("There
are")queue-callswaiting = "queue-callswaiting" ; ("calls
waiting.")queue-holdtime = "queue-holdtime" ; ("The current est. holdtime
is")queue-minutes = "queue-minutes" ; ("minutes.")queue-thankyou =
"queue-thankyou" ; ("Thank you for your patience.")
And have verified that the files exist, with proper
permissions (644), in /var/lib/asterisk/sounds However, when a call
gets dumped in the queue, the console shows the following:
Nov 20 16:40:32 WARNING[1120924864]: file.c:475
ast_openstream: File "queue-youarenext" does not exist in any formatNov 20
16:40:32 WARNING[1120924864]: file.c:779 ast_streamfile: Unable to open
"queue-youarenext" (format GSM): No such file or directory
Do queue sound files get pulled from a different
location?
Suggestions?
Thanks!
Andy
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[Asterisk-Users] ANY DEVELOPERS HERE? "warning: implicit declaration of function `__use_ast_pthread_create_instead__"
I have been running a version of Asterisk that is 4-5 months old. When trying to upgrade to 1.0.2, I get several compile warnings such as: chan_zap.c:3515: warning: implicit declaration of function `__use_ast_pthread_create_instead__' The channel modules will not load with the error: undefined symbol: __use_ast_pthread_create_instead__ I have removed the modules before compiling, make clean, make install, reinstalled them, did the modprobe stuff, etc. Do I have some type of missing library or function that is causing the ast_pthread errors? I got the Asterisk, Zaptel and Libpri off the CVS. It looks like my gcc is current. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] A new alternative to see who is online
Is there any chance that you might release the source code so that others can improve upon your code? I can see a real need for an application like this. I just wish it could be tweaked a little. -- Jim Dossey Computer Services -- Original message -- From: [EMAIL PROTECTED] > Hi all, > > I have been facing about the problem to know who is online with asterisk PBX. > However users wanted to see it right away, without launching any application. > As > I could not find any solution with IP phones and users were really > complaining, > I decided to write this little application that runs under windows and stays > on > screen. > > It is not perfect, but it works and I think it can help other people. > > Have a look: > http://mapage.noos.fr/~b.nico/ > > Regards, > Nicolas > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phones-Receptionist Setup
Title: SIP Phones-Receptionist Setup I am looking at placing a system in an office with a central receptionist, and phones for each individual employee thereafter. Could I use a Snom 220 with additional keypads to view if the lines are in use by the other employees? Fred is in sales... A call comes into the receptionist and they transfer the call to Fred. The receptionist can tell Fred is still on the phone by viewing the assigned key on the Snom 220’s keypad, so if another call comes in they know he is on the phone instead of just blindly transferring the call and pushing the person to his voicemail. So they can ask the person hold or if them want to be transferred into Fred’s voicemail. Any thoughts would be appreciated... I am new to the list, so my apologies if this has been addressed before. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Dialstatus
Hello,
I've got some SIP clients, and an IAX2 long distance provider. Ideally,
when a the dialed number is busy I will hear a busy signal. Instead, I
get Congestion even though * knows it's busy. Is this a bug or am I
missing something?
The dial plan, in basically this
Dial(IAX2/[EMAIL PROTECTED]/19995551234,,)
Goto(failedcall-${DIALSTATUS})
failedcall-CONGESTION plays congestion
failedcall-CHANUNAVAIL plays congestion
failedcall-BUSY plays busy
everything else loops back to congestion
Here's the output from the * console, with most provate data
search/replaced.
-- Executing Dial("SIP/dc1-aa6a", "IAX2/[EMAIL PROTECTED]/19995551234")
in new stack
-- Called [EMAIL PROTECTED]/19995551234
-- Call accepted by 1.2.3.4 (format ULAW)
-- Format for call is ULAW
-- IAX2/provider/1 is making progress passing it to SIP/dc1-aa6a
-- IAX2/provider/1 is busy
-- Hungup 'IAX2/provider/1'
== Everyone is busy/congested at this time
-- Executing Goto("SIP/dc1-aa6a", "failedcall-CHANUNAVAIL|1") in new
stack
-- Goto (trevsip,failedcall-CHANUNAVAIL,1)
-- Executing Congestion("SIP/dc1-aa6a", "") in new stack
== Spawn extension (trevsip, failedcall-CONGESTION, 1) exited non-zero
on 'SIP/dc1-aa6a'
Any hints would be appreciated...
Thanks,
Trevor Peirce
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[Asterisk-Users] SIP Call not Approved
Hello I am getting message "Call not Approved" while using xlite SIP phone. Phone is registered correctly and I am able to get incoming call but on out going calls I am getting "Call Not Approved" message. Please through some ideas. Thanks Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX issue at nufone
Hello: . I'm having troubles registering on nufone's IAX service . I'm really new to Asterisk Nufone provide me some config examples ... I can dialout but I can't register my * Box, eg. whe I do "iax show registry" I got only a "Request Sent" and later I have a "Timeout" My box is on a Public IP and no firewall. I'm using *0.5 on a FreeBSD 5.2 box and the sip part is working fine Any ideas would be appreciated Thanks in advance, and excuse me by my english :o( config from nufone - Your number is: 888-899-. Add the following to your iax.conf: [general] register => user:[EMAIL PROTECTED] [NuFone] <-- yes you can have two of the same [name] type=user secret=passwd context=inbound and your extensions.conf: [inbound] exten => 99,1,Answer --- Ing. Julio Alvarez Tejera Unix Trends *BSD, Solaris & Linux --- "extremely stable systems" ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] How to encript SIP comunications?
Hello Fach, I have used openvpn for a while and in the new release thereis a feature called "server mode" that makes posible to have a full network of vpn links besides a single TUN/TAP adaptor (a pure software NIC) in the server. I haven't used that feature, but I think this is what you need. Also openvpn runs on linux, *bsd, solaris, windows, and maybe in other OS. Miguel > >Hello Gregory > >Thanks for your tip, but this looks like a point to point encription, >but how about between extensions registered in a Asterisk server. > >Let's say I got a building 200 users registered and a given set of >extensions, any of the users can be out of town or in another building >in another city but for the matter of their job their communications >have to be encripted. I can do your suggestion, but is group of users >move from place to place then how would I do? > >I would appreciate to have a clear solutions for a more flexible >scenario of encription > >All suggestions are highly appreciated > >Bye > >Fach = Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to encript SIP comunications?
Am Samstag, 20. November 2004 18:48 schrieb Linux Dominicana: > Hello Gregory > > Thanks for your tip, but this looks like a point to point encription, > but how about between extensions registered in a Asterisk server. > > Let's say I got a building 200 users registered and a given set of > extensions, any of the users can be out of town or in another building > in another city but for the matter of their job their communications > have to be encripted. I can do your suggestion, but is group of users > move from place to place then how would I do? > > I would appreciate to have a clear solutions for a more flexible > scenario of encription > > All suggestions are highly appreciated > Hi all I did some research on this topic a short time ago. Here is what's the status. Anyone correct me, if I'm wrong: Encryption can be done on several OSI Layers Layer 3: IPsec (Network Layer) The network layer secures the connection. Unfortunately you have to use an up to date kernel and hard phones don't support it yet. This sounds pretty secure but can only be done in point to point so here we have our disadvantage for your case. Layer 4: TLS ( Transport Layer) Hell I simply forgot what was wrong here. I think it was the NAT traversal. You can not secure at this layer, if you have a NAT between your boxes because the checksum has to be altered for changing IP's. I'm not quite sure about this as I said I simply forgot or I'm mismixing layers :-). Layer 5/6 : SRTP Security at the application level. Well this is what we want. SRTP is defined in an rfc and it secures any stream. A reference implementation was created by a big company which does nothing more than wrap a security layer around the RTP protocol. This is unfortunately just the half lease because it only secures the audio stream. We would also want to secure the signaling protocol which we call SIP. This is called SIPS and is also defined in a rfc. So now we know what we want but how do we want it? There is symmetric encryption (same key used for encryption as for decryption) and there is asymmetric encryption (different keys for encryption and decryption). Symmetric ciphers are DES, AES, 3DES and so on. These are good for real-time applications such as voice audio as they are fast enough to en/decrypt lots of data in a short period of time. Well asymmetric encryption is mainly done via RSA-based ciphers which are quite hard to handle in large-scale environments for many reasons especially key exchange can get complex for software and hardware. So some vendors sell phones that support AES encryption. SNOM claimed to support it but has removed this support for some reason from their data sheets. The Zip4x5 claimes to have encryption and you can even download the software for linux for free if you are willing to give your name and email away. Well and now we finally get to the problem: Asterisk is somehow ready to support encryption as AES libs are compiled in but there is no SRTP and SIPS protocol implementation as far as I could see. Someone on IRC told me that he thinks encryption will be done in about a half year but my personal estimation regarding the latest development of asterisk would be a bit longer as a lot of things have to be reworked from scratch. Due to the frequent patches and contributions to the code from many developers the code gets more and more messed up. I haven't checked the development version for quite a while so this might have changed in the meantime. Anyone feel free to post it to the wiki if you like. So far Jens ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Shared line appearances
Maybe a bounty would help. I'd be willing to chip in. John Baker Paul Rodan wrote: I don’t think the nature of these phones would allow for such a thing. It was designed for transfers and such, to be a real PBX, not like having 4 phone lines from BellSouth and multiple 4 line phones. I couldn’t imagine SER/Asterisk/any SIP proxy or program doing what is needed. The only idea I had to get asterisk to do it would be have the calling party thrown into a conference room right away, and then have it ring all the other phones. Whoever answers it would then be put into the conference room with the calling party. But I think the trick is, whenever a person calls in, they get thrown put into a conference room, and then the PolyCom’s all have to auto-answer and place the calls on silent hold, so that everybody is thrown into the conference room. That shouldn’t be TOO hard to rig, but how do you get all the phones to ring as well until somebody picks up, so that there is at least 1 active person in the conference with the calling party. Then any other phone should be able to bust in simply by taking that line off of hold. Good luck with that J *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *David Gomillion *Sent:* Friday, November 19, 2004 12:19 PM *To:* [EMAIL PROTECTED] *Subject:* [Asterisk-Users] Shared line appearances All right. It would appear that I am not the only one interested in shared line appearances. Many others have stated that they wish for the key-system-like feature of the blinking lights. Quite frankly, I don't think it's a good thing, but the people who use these systems are very resistant to change. I set up call queues, pickup groups, 1-touch transfers, and still nothing seems to placate them. If I could, I would just replace the users... I mainly use Polycom SoundPoint IP phones: some 300s and some 600s. Bottom line is that if I am going to be able to finish the rollout of the phone system, and switch away from having 2 PBXs vying for power (Asterisk and the Nortel NorStar MICS system), I am going to have to get this feature working. I have received authorization to offer a bounty to get it working in Asterisk, and to then contribute the source to the project. As I have studied the issue, I'm not sure it is within the "master plan" for asterisk. Searching the archives, it seems we only expect Asterisk to be a "clever UA". The people asking were advised to get a real SIP proxy. In passing, someone asked if chan_sip2 would support it, but I found no response. Many references to SER have been made. I have installed SER successfully. I then tried to make the feature work, but have been unsuccessful. Both lines will ring, but the first person to answer the call gets it, and the other phone's lights are as dark as can be. SER does not seem to do any better with "line-seize" than Asterisk. At least Asterisk has the hint to allow the lights to work (I have not yet implemented this, but since it does not meet the requirements, it does not really matter)... but neither system will allow the caller to press the blinking light on a call that was placed on hold to answer it. I am now looking at other SIP proxies. I am in the process of installing sipXpbx, which includes many different pieces of the Pingtel sipExpress system that have been open-sourced. I am not sure which pieces I will need specifically, so I will install the whole shooting match and see if the feature even works. If it does, I'll remove packages and try to reintegrate with Asterisk. Has anyone gotten shared line appearances to work with Polycom Soundpoint IP phones? Not just blinking lights, but the whole shebang: lights, pressing the button to seize the line, shared registrations, etc. Is it best to work with a 3rd party SIP proxy/router/whatever, or should we pool resources and get the feature integrated into Asterisk somehow? Looking forward to your thoughts, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Multiple asterisk process
David Boyd wrote: [snip] > > Greg, you need chill; take a deep breath; now say to yourself, let it > g!! > Does "hypertensio arteriale" and "myocardial infarction" ring a bell..? > Critch, has the right to respond, anyway he desires. People need to > be responsible for themselves and their actions, and in particular > they need to defend themselves if they feel attacked or insulted. > > I have not seen a response from the individual who posted the original > question (Hong) reply at all to the thread; if he isn't concerned then > why are you? > He probably resigned from the maillinglist screaming... > Why do you think the list as a whole reflects something about you, > only your posts say anything about you. > > I don't wonder at all about Linux catching on, it is, one informed > user at a time! > Can we conclude that the following chinese proverb still is valid ?? "He who asks may be a fool for five minutes. But he who does not ask remains a fool forever." On the other hand... This may be more appropriate... "Accept that some days you're the pigeon, and some days you're the statue." Scott Adams. :-) /Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multiple asterisk process
On 10:42 AM 11/20/2004, Jose Hernandez wrote: > >>Did you bother using google? > >I searched google but could not find an answer. Any other suggestions? > http://lists.digium.com/pipermail/asterisk-users/2004-April/043852.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playing announcement when call is answered
Okay, I posted a question earlier asking how to start the monitor
command when the call is answered not when it is dialed. I managed to
fix the problem myself by using a macro in the Dial string like this:
exten => 21060,1,Dial(${Nick},20,M(monitor))
Next problem though! What I need to do is play a legal announcement to
the caller when the the call is answered. Now I thought this would be
easy, but it seems I am missing something. I simply added the playback
line to the same macro that I used for my monitor command.
I made sure the call was answered before commencing playback along
with having a pause and removed the r option from the dial string so
as to not block playback on the line. But when I call in the audio is
played on the destination extension but not to the caller. Caller just
hears the ring until the macro is complete then conversation
commences. My macro I am using is below.
[macro-monitor]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Playback(legal-announcement)
exten => s,4,Monitor(wav49,${UNIQUEID},m)
I have tried lots of variations of this, including using Background
not Playback.
Anyone have any ideas?
TIA for any help anyone can offer.
Oh yea, I also tried answering the call in the dial string rather than
the macro but this had the same effect. I also want to avoid answering
the call before the call has been picked up, as this will be an
extension of an existing PABX, so we want the call to return to
reception if the call goes unanswered. Not sure how I would go abouts
doing this if astrisk answers the call?
The test system I am doing this on has calls incoming via IAX and
terminating on SIP extensions. But the final production system will be
using Zaptel hardware for the incoming call and SIP extensions.
Cheers
Nick
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Re: [Asterisk-Users] Codec negotiation
On Sat, 2004-11-20 at 18:48 +0100, Tamas J wrote: > Hello! > > I would like to know wether it is possible to have end-to-end codec > negotiation in iax2? > What I mean is... > > In case the user dials a number available through PSTN, let's force to > use alaw (the client is in LAN) to overcome unneeded transcoding: > iaxphone->1st asterisk -> PSTN > > In case the same user dials a number available throug a chain of IAX2 > peers (e.g. 2 peers), try to negotiate the codec end-to-end to consume > less resources for transcoding on asterisk servers (of course, in that > case we don't want to use g711, but ilbc, speex or gsm). > iaxphone->1st asterisk->2nd asterisk->PSTN > Or maybe: > iaxphone->1st asterisk->2nd asterisk->iaxpohone > > Is there a way to do that? If yes, how? If 1st asterisk -> 2nd asterisk is a link that negotiates the ILBC, gsm, or speex, when the call transfers, it should negotiate the codec. Of course part of the interesting effect here is that unless there is NAT or something similar in the way, IAX is going to try and get out of each section if it can. So you may end up with the end result being iaxphone -> iaxphone and they might be negotiating with each other. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec negotiation
Hello! I would like to know wether it is possible to have end-to-end codec negotiation in iax2? What I mean is... In case the user dials a number available through PSTN, let's force to use alaw (the client is in LAN) to overcome unneeded transcoding: iaxphone->1st asterisk -> PSTN In case the same user dials a number available throug a chain of IAX2 peers (e.g. 2 peers), try to negotiate the codec end-to-end to consume less resources for transcoding on asterisk servers (of course, in that case we don't want to use g711, but ilbc, speex or gsm). iaxphone->1st asterisk->2nd asterisk->PSTN Or maybe: iaxphone->1st asterisk->2nd asterisk->iaxpohone Is there a way to do that? If yes, how? Thanks in advance, TamasJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to encript SIP comunications?
Hello Gregory Thanks for your tip, but this looks like a point to point encription, but how about between extensions registered in a Asterisk server. Let's say I got a building 200 users registered and a given set of extensions, any of the users can be out of town or in another building in another city but for the matter of their job their communications have to be encripted. I can do your suggestion, but is group of users move from place to place then how would I do? I would appreciate to have a clear solutions for a more flexible scenario of encription All suggestions are highly appreciated Bye Fach On Sat, 20 Nov 2004 00:39:28 -0500, Gregory Junker <[EMAIL PROTECTED]> wrote: > Linux 2.6 kernel includes IPSec directly, and ipsec-tools can be used to > create a secure point-to-point link. OpenSWAN makes use of the kernel > IPSec in 2.6, and makes it available in 2.2 and 2.4 kernels. IPSec can > use shared keys or x509 certificates within or without a PKI for > authentication. OpenVPN has been mentioned as another option, and it > uses SSL/TLS for the encryption, and also supports PKI and PSK for auth. > Both provide perfect-forward secrecy (PFS) which is important if your > client wants past and future communications to remain impossible to > decrypt, even with a compromised or subpoenaed private key. > > Any of the above can be used to encrypt a point-to-point link such as > the one you describe. > > http://www.openswan.org > http://www.openvpn.org > > Greg > > > > Linux Dominicana wrote: > > Hello everybody > > > > A given scenario: > > > > A client does want to have his own VoIP PBX with Asterisk running, but > > he ask me. How secure can be the communication among all subscribers? > > If there're sniffers on the middle or any other listening device on a > > given netowork. > > > > The client is not fictitial, but it main requirement is encription of > > all point to point comunications for given reasons. > > > > Any guidance, products, solutions implementation available and if > > works is much better. > > > > Suggestions are welcome > > > > Regards > > > > John Fach > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- John Fach Linux Dominicana Linux/LAMP Technology Consulting & Solutions p: 1-786-380-4685 1-347-952-3288 w: http://www.linuxdominicana.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can anyone shed some light on wht these calls were dropped?
Hi,
I need help finding why my system is dropping calls..
I enabled debugging on my box in the hope it would lead me to the answer
as to why my system is dropping calls but unfortunately nothing is
jumping out at me..
I have attached the portion of the messages file for two calls that were
dropped.. (numbers names and ip's have been changes to protect the innocent)
Can someone give me a reason why the calls were dropped? or maybe tell
me where to look
My only thought is that its a loosing some packets as it goes across the
internet to the IAX provider and that the tolerance for packet loss is
set really low so it drops the call completely..
Any help appreciated..
Thanks..
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 1 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:00 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:01 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 9 on Zap/2-1
Nov 20 16:52:02 DEBUG[-1109894224]: DTMF digit: 0 on Zap/2-1
Nov 20 16:52:05 DEBUG[-1109894224]: Enabled echo cancellation on channel 2
Nov 20 16:52:05 DEBUG[-1109894224]: Made call 5 into trunk call 16384
Nov 20 16:52:05 DEBUG[-1109894224]: Created trunk peer for
'111.222.333.444:4569'
Nov 20 16:52:05 DEBUG[-1109894224]: Expanded trunk '111.222.333.444:4569' to
6400 bytes
Nov 20 16:52:07 DEBUG[-1109894224]: Received AST_CONTROL_PROGRESS on Zap/2-1
Nov 20 16:52:07 DEBUG[-1095144528]: Ooh, voice format changed to 1024
Nov 20 16:52:15 DEBUG[-1109894224]: Took Zap/2-1 off hook
Nov 20 16:53:35 DEBUG[-1090942032]: Setting NAT on RTP to 0
Nov 20 16:53:36 DEBUG[-1090942032]: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Found
Nov 20 16:53:48 DEBUG[-1090942032]: Auto destroying call '[EMAIL PROTECTED]'
Nov 20 16:55:01 DEBUG[-1109894224]: Exception on 17, channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Got event Wink/Flash(3) on channel 2 (index
0)
Nov 20 16:55:01 DEBUG[-1109894224]: Winkflash, index: 0, normal: 17, callwait:
-1, thirdcall: -1
Nov 20 16:55:01 DEBUG[-1109894224]: Already have a dsp on Zap/2-2?
Nov 20 16:55:01 DEBUG[-1109894224]: Swapping 2 and 0
Nov 20 16:55:01 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:01 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:04 DEBUG[-995472]: Exception on 17, channel 2
Nov 20 16:55:04 DEBUG[-995472]: Got event On hook(1) on channel 2 (index 0)
Nov 20 16:55:04 DEBUG[-995472]: Last flash was 2764 ms ago
Nov 20 16:55:04 DEBUG[-995472]: Swapping 2 and 0
Nov 20 16:55:05 DEBUG[-995472]: disabled echo cancellation on channel 2
Nov 20 16:55:05 DEBUG[-995472]: waitfordigit returned < 0...
Nov 20 16:55:05 DEBUG[-995472]: Hangup: channel: 2 index = 2, normal = 17,
callwait = -1, thirdcall = 21
Nov 20 16:55:05 DEBUG[-995472]: Released sub 2 of channel 2
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 2, ts=160005,
seqno=47)
Nov 20 16:55:06 WARNING[-1095144528]: Max retries exceeded to host
111.222.333.444 on IAX2/provider-out/16384 (type = 6, subclass = 11, ts=160008,
seqno=48)
Nov 20 16:55:06 DEBUG[-1109894224]: Didn't get a frame from channel:
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: Bridge stops bridging channels Zap/2-1 and
IAX2/provider-out/16384
Nov 20 16:55:06 DEBUG[-1109894224]: We're hanging up IAX2/provider-out/16384
now...
Nov 20 16:55:06 DEBUG[-1109894224]: Exiting with DIALSTATUS=ANSWER.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: inserting a CDR record.
Nov 20 16:55:06 DEBUG[-1109894224]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2004-11-20 16:52:05','\"Cordless Phone\"
<1234>','1234','90019090909090','local',
'Zap/2-1','IAX2/provider-out/16384','Dial','IAX2/[EMAIL
PROTECTED]/19058289575',181,171,'ANSWERED',3,'Cordless Phone')
Nov 20 16:55:06 DEBUG[-1109894224]: Hangup: channel: 2 index = 0, normal = 17,
callwait = -1, thirdcall = -1
Nov 20 16:55:06 DEBUG[-1109894224]: disabled echo cancellation on channel 2
Nov 20 16:55:06 DEBUG[-1109894224]: Set option TDD MODE, value: OFF(0) on
Zap/2-1
Nov 20 16:55:06 DEBUG[-1109894224]: Updated conferencing on 2, with 0
conference users
Nov 20 16:55:08 DEBUG[-1095144528]: Immediately destroying 1638
Re: [Asterisk-Users] Need help selecting phones
Take a look at cisco phones http://www.cisco.com/en/US/products/hw/phones/ps379/ and snom http://www.snom.com/products_de.htm and wiki http://www.voip-info.org/wiki-Asterisk+phones I'm new to the asterisk world and have been playing with an asterisk server with 1 FXO card for a couple of weeks. Now I'm looking to start adding IP Desk Phones but I'm unable to come to a decision on what phones to use. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] A new alternative to see who is online
[EMAIL PROTECTED] wrote: Hi all, I have been facing about the problem to know who is online with asterisk PBX. However users wanted to see it right away, without launching any application. As I could not find any solution with IP phones and users were really complaining, I decided to write this little application that runs under windows and stays on screen. Nicolas - Good work with Leon! Could I just make one feature request? How about the ability to tuck the application into the system tray, so you could just click its icon and have a vertical display pop out. Kind of like the GNOME Clock (see http://www.not-real.org/scr.html if you're not familiar)... I think this would be handy for those users who don't like a cluttered desktop... Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A new alternative to see who is online
Hi all, I have been facing about the problem to know who is online with asterisk PBX. However users wanted to see it right away, without launching any application. As I could not find any solution with IP phones and users were really complaining, I decided to write this little application that runs under windows and stays on screen. It is not perfect, but it works and I think it can help other people. Have a look: http://mapage.noos.fr/~b.nico/ Regards, Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: Multiple asterisk process
Tom Ivar Helbekkmo wrote: Matt Riddell <[EMAIL PROTECTED]> writes: 3. Maybe a new rule :-) Don't flame, but if someone does, don't respond to it. It will go away if you ignore it. Better yet, don't respond in public. If you disagree with technical content, say so on the list, but if you object to the form of a message, tell the author in private email. Whoa!!! Isn't public humiliation part of the reason for flaming someone? Why would one do it in private, and thereby risk being perceived as kind and helpful? You gotta knock these newbies around in *public* folks; the whole point is for people who are in the know to make sure the newbies are put in their place. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] three way mixing / conferencing
m. smadi wrote: There are lots of sip phones which do not support conference calling and I would like to know if asterisk can support that (i.e. perform the mixing) on behalf of the phone? For example if RTP A <-- * --> B and then C tries to call B where the B phone can support one call at a time, would * support: Check the wiki for MeetMe and/or app_conference. Asterisk can conference as many streams as you wish, but making it work seamlessly with your SIP phones may be a bit tricky. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] block caller id
Brancaleoni Matteo wrote: I think that doing SetCIDNum() (with no args) before dial will do the trick. Depends on your version of Asterisk. Newer versions have a CallingPres() app that you can use to specify the presentation handling of your CLID and CNAM information when you send it to the telco. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] zaphfc sound problems
Hi I recommend you abandon the old card with the CAPI drivers and purchase a second HFC card. You can get them for around £10/EUR15 approx if you shop around on the net. You need to use zaphfc and the native zaptel and libpri drivers. You will need the bristuff from http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-rc2b.tar.gz which has a script to download and install the whole thing automatically. The HFC card will generate 8000 interrupts per second. it is normal operation. Do make sure you are not sharing these IRQs or you will have problems. I have 3 HFC cards running in an Celeron 400. Rgds Tim Thomas Jagoditsch wrote: hi list. after my unsuccessfull experiences with mISDN i tried again to implement a zaphfc based solution. problem is: sound on calls via capi is stuttering/broken and therefore unuseable. my conf: - cel 1300, 256mb ram - avm b1 via capi connected to my outgoing ISDN - acer surf pci via zaphafc and crosslink cable with termination as internal bus - siemens gigaset 3035 as internal phone - various sip-phones on my pc to test software versions: - gentoo with kernel 2.6.9 - bri-stuff.0.1.0-RC4a installed with cvs -z9 co -D '2004/08/21 21:34:58' asterisk libpri zaptel all worked fine and this was the first time i could get the gigaset to talk to asterisk (in contrast to mISDN). but ... the sound problem: ive got stuttering noises, voice is almost not understandable in some cases - but in every case ;-( symptoms: - calls form gigaset to asterisk (demo,menu etc) are ok - calls from gigaset via asterisk to pstn have stuttering - calls from sip via asterisk to pstn are ok - calls from pstn to asterisk (demo, menu etc) have stuttering - calls from pstn via asterisk to gigaset have stuttering - calls from pstn via asterisk to sip are ok what me stikes here is that you cant find a definitive suspect here. zaphfc itself seems to be ok - demo and menu sound are ok and otoh i get problems when zaphfc is out of the game - like calling asterisk alone from pstn. but capi itself works - at least when i use sip to call out. ive got to stop scratching my head about this before i loose to much hair ... i googled around and found no solution. what i tried: - the hdparm thing - switching apci/acpi - changing resitors for termination of the internal bus (100/50 ohms) or remove termination completely one thing that makes me curious are interrupts: Sat Nov 20 13:49:30 CET 2004 CPU0 0:4270237 XT-PIC timer 1: 44 XT-PIC i8042 2: 0 XT-PIC cascade 5: 32813694 XT-PIC zaphfc 8: 2 XT-PIC rtc 10: 16843 XT-PIC b1pci-b000 11: 16563 XT-PIC uhci_hcd, uhci_hcd, CMI8738, eth0 14: 1822 XT-PIC ide0 15: 17 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Sat Nov 20 13:49:40 CET 2004 CPU0 0:4280248 XT-PIC timer 1: 44 XT-PIC i8042 2: 0 XT-PIC cascade 5: 32893770 XT-PIC zaphfc 8: 2 XT-PIC rtc 10: 16844 XT-PIC b1pci-b000 11: 16569 XT-PIC uhci_hcd, uhci_hcd, CMI8738, eth0 14: 1822 XT-PIC ide0 15: 17 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 there was no activity on the system and zaphfc produces interrupts like crazy. i found one post describing the same thing - but the thread concludes without solution ... someone got ideas ? or maybe i could try another approach: has anyone got a similar working configuration (isdn external & internal via capi and/or zaphfc and/or mISDN) ? which hardware / software do it run on ? wbr.tja... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] three way mixing / conferencing
There are lots of sip phones which do not support conference calling and I would like to know if asterisk can support that (i.e. perform the mixing) on behalf of the phone? For example if RTP A <-- * --> B and then C tries to call B where the B phone can support one call at a time, would * support: C \ \ \ RTP \ * --> B / / / RTP / A thanks moe smadi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting the EXTEN variable - is it possible?
Hi! Now I think I know how to handle AGI from PHP. I'm about to write an AGI-script that enables me to dial numbers of different lengths. I've got a PhoneJack and the "chan_phone" releases the digits directly when entered. So it isn't possible to define dialplans with different lengths of digits. Now I wrote a PHP script. It isn't perfect by now. But when it is it will not only have a normal timeout. But the timeout will be calculated depending on the input speed of the user. Now its the question what to do with the number. I can program a dialplan in PHP. That's no problem. But I want to see if its possible to use the standard dialplan as defined in the extension.conf. The dialed extensions is in the global variable "EXTEN". But it seems to me as if it isn't possible to manually set this variable and rerun the dialplan? Or is there any way to do so? Thanks! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to encript SIP comunications?
Linux Dominicana wrote: Hello everybody A given scenario: A client does want to have his own VoIP PBX with Asterisk running, but he ask me. How secure can be the communication among all subscribers? If there're sniffers on the middle or any other listening device on a given netowork. The client is not fictitial, but it main requirement is encription of all point to point comunications for given reasons. Any guidance, products, solutions implementation available and if works is much better. There is SRTP: http://srtp.sourceforge.net/srtp.html Doesn't look like anyone has submitted a feature request for its inclusion in *. Regards, -- Jason Becker Director & CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax testing using loop-back
I'm writing a fax server using Steve Underwood's spandsp package. I have email to fax working. I can send an email with pdf/ps attachments to [EMAIL PROTECTED], and * will TxSend the content to the fax machine on that number (my internal qmail server forwards .fax domains to the qmail server on the */fax server). I also have fax to email working. I can fax to 3036701917, a line on the */fax server, and it will detect and TxRecv the fax and then email the content as a pdf attachment back to me. So, to save a little paper, I would like to TxSend out on one * line to a different * line which will TxRecv (a loop back scenario). Using a butt set, I can hear the two lines connect and the negotiation tones between the two "modems". I can then hear about one second of white noise, (which I believe is hdlc data), and, after that, one "modem" becomes silent while the other sends negotiation tones (forever). I have to manually unplug the lines from the TDM card to reset. In my call file, I do have "Data /tmp/file.tiff|caller" specified. Does anyone have this working? Thanks, Mike P.S. Does anyone know of a MS Word to PDF/PS converter that runs on Linux? -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc sound problems
hi list. after my unsuccessfull experiences with mISDN i tried again to implement a zaphfc based solution. problem is: sound on calls via capi is stuttering/broken and therefore unuseable. my conf: - cel 1300, 256mb ram - avm b1 via capi connected to my outgoing ISDN - acer surf pci via zaphafc and crosslink cable with termination as internal bus - siemens gigaset 3035 as internal phone - various sip-phones on my pc to test software versions: - gentoo with kernel 2.6.9 - bri-stuff.0.1.0-RC4a installed with cvs -z9 co -D '2004/08/21 21:34:58' asterisk libpri zaptel all worked fine and this was the first time i could get the gigaset to talk to asterisk (in contrast to mISDN). but ... the sound problem: ive got stuttering noises, voice is almost not understandable in some cases - but in every case ;-( symptoms: - calls form gigaset to asterisk (demo,menu etc) are ok - calls from gigaset via asterisk to pstn have stuttering - calls from sip via asterisk to pstn are ok - calls from pstn to asterisk (demo, menu etc) have stuttering - calls from pstn via asterisk to gigaset have stuttering - calls from pstn via asterisk to sip are ok what me stikes here is that you cant find a definitive suspect here. zaphfc itself seems to be ok - demo and menu sound are ok and otoh i get problems when zaphfc is out of the game - like calling asterisk alone from pstn. but capi itself works - at least when i use sip to call out. ive got to stop scratching my head about this before i loose to much hair ... i googled around and found no solution. what i tried: - the hdparm thing - switching apci/acpi - changing resitors for termination of the internal bus (100/50 ohms) or remove termination completely one thing that makes me curious are interrupts: Sat Nov 20 13:49:30 CET 2004 CPU0 0:4270237 XT-PIC timer 1: 44 XT-PIC i8042 2: 0 XT-PIC cascade 5: 32813694 XT-PIC zaphfc 8: 2 XT-PIC rtc 10: 16843 XT-PIC b1pci-b000 11: 16563 XT-PIC uhci_hcd, uhci_hcd, CMI8738, eth0 14: 1822 XT-PIC ide0 15: 17 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Sat Nov 20 13:49:40 CET 2004 CPU0 0:4280248 XT-PIC timer 1: 44 XT-PIC i8042 2: 0 XT-PIC cascade 5: 32893770 XT-PIC zaphfc 8: 2 XT-PIC rtc 10: 16844 XT-PIC b1pci-b000 11: 16569 XT-PIC uhci_hcd, uhci_hcd, CMI8738, eth0 14: 1822 XT-PIC ide0 15: 17 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 there was no activity on the system and zaphfc produces interrupts like crazy. i found one post describing the same thing - but the thread concludes without solution ... someone got ideas ? or maybe i could try another approach: has anyone got a similar working configuration (isdn external & internal via capi and/or zaphfc and/or mISDN) ? which hardware / software do it run on ? wbr.tja... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multiple asterisk process
On Sat, 2004-11-20 at 01:32, Gregory Junker wrote: > > Add to it, my message wasn't a flame but rather a terse comment. Your > > Never said it was a flame. I said it was in a tone virutally guaranteed > to make the guy consider you and everyone on the list to be a conceited > jackass. > > The difference in your perception of your replies (the one I snipped > included) and the way you actually come off in public, is the problem. > You think you are being terse. You actually thought your post directed > the guy to the "answer repository". He probably did end up going to > Google, but I'll bet he loses interest in Asterisk before long. I guess > your work is done here then, right? If they guy isn't an expert, he has > no hope of learning, huh? > > And they wonder why Linux doesn't catch on... > > > Greg > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Greg, you need chill; take a deep breath; now say to yourself, let it g!! Critch, has the right to respond, anyway he desires. People need to be responsible for themselves and their actions, and in particular they need to defend themselves if they feel attacked or insulted. I have not seen a response from the individual who posted the original question (Hong) reply at all to the thread; if he isn't concerned then why are you? Why do you think the list as a whole reflects something about you, only your posts say anything about you. I don't wonder at all about Linux catching on, it is, one informed user at a time! Dave P.s. Sorry for bottom posting in my reply;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Starting AGI when handset is picked up?
Michael Vogel schrieb: Now I have got to find out how to make AGI play the dialtone until a digit is entered. I found several commands like "Playtones" but it doesn't work ... Now it works. I'd only got some problems using parameters when calling external applications. But now it seems that I can do everything I wanted to do. Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Multiple asterisk process
Matt Riddell <[EMAIL PROTECTED]> writes: > 3. Maybe a new rule :-) Don't flame, but if someone does, don't respond > to it. It will go away if you ignore it. Better yet, don't respond in public. If you disagree with technical content, say so on the list, but if you object to the form of a message, tell the author in private email. -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Need help selecting phones
check out the snom 190. I am also in the process of making the 3com take a sip image and testing with asterisk. both are pricey but look very good on a corp. desk. - Original Message - From: Peter Awad To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, November 19, 2004 8:56 AM Subject: [Asterisk-Users] Need help selecting phones Im new to the asterisk world and have been playing with an asterisk server with 1 FXO card for a couple of weeks. Now Im looking to start adding IP Desk Phones but Im unable to come to a decision on what phones to use. I like to look of the Polycoms, but because we are not a phone company I cant see us getting reseller authorized for them. Shoretel has some nice looking phones, but I dont want to be forced into buys their PBXs as well. I dont like to look of the grandstream budgetel stuff as it looks like its name implies. I would really like SIP, multi-line display, multiple extensions, and handsfree. Can someone recommend a line of phones that work well with * and are distributed in Canada? Id prefer a distributor located in Canada but that is not a priority over getting a good line of phones that I fell we can put side by side any digital system and say we can do everything that phone can do. Any recommendations would be great. Thanks Peter ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Variable Codecs Order
Yes , for iaz
but what if the originator and destination are SIP and are registerd in
the local * box.
How to create peer for local * ?
Chris Stenton wrote:
setup a seperate context for each either in iax.conf or sip.conf
ie extensions.conf
IAX2/[EMAIL PROTECTED]/${EXTEN:1}
iax2.conf
[ATOB]
host=pbx.atob.org
disallow=all;
allow=ulaw;
etc
Chris
- Original Message - From: "Miroslav Nachev"
<[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 28, 2004 4:37 PM
Subject: [Asterisk-Users] Variable Codecs Order
Hi,
Is it possible the codecs order to be different depending of
originator and destination? For example:
1. A to B: G.729, G.723, G.711
2. A to C: G.711
3. B to C: G.711
4. A to D: G.723, G.729
Any ideas?
--
Best Regards,
Miroslav Nachev
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--
Best Regards,
Damian Minkov
COSMOS Software Enterprises, Ltd.
Tel:(+359-2) 983-32-62
Mobile: (+359-88) 853-28-25
E-Mail: [EMAIL PROTECTED]
http://www.space-comm.com
Post address:
P. O. Box 941,
1000 Sofia,
Bulgaria
Office address:
ap. 9, fl. 4,
"11 August" str., No. 43,
1202 Sofia,
Bulgaria
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[Asterisk-Users] SipTone II
Anybody used the above phone with asterisk
I have one working ok for calls, but having a problem with voice mail.
Using either the 'Voice mail function key' or dialing 88 (for my system)
just gets me to Call Terminated
Asterisk CLI shows the error message 'unable to get User name'
My Grandstream works ok, asking for User name, then Password
Any ideas ?
Hi Clive,
I've been using a SipTone II for quite a while. Great phone but kind of
pricey.
I got the VM key working by configuring the Voicemail Server item
in the Phone Configuration web interface section as follows:
sip:[EMAIL PROTECTED]
where voicemailextension is the extension number for accessing
voicemail in * and asterisk.company.com is the domain name or
IP address of your * machine.
I'm using Firmware version: SipTone 1.2.0 rc Z_8.
Hope this helps.
Michael Swan
Thanks for your reply Michael. I don't think I explained my problem very
well.
I was dialling '88' (my voicemail extension) from the Siptone to get to
Voicemail.
I have tried your tip , but with the same result
The CLI shows
Executing VoiceMailMain("SIP/2004-43c0","") in new stack
Playing 'vm-login'
WARNING [655381] app-voicemail.c:2748 vn-execmain : Couldnt read Username
Spawn extension (internal,88,2) exited non-zero on 'SIP/2004 - 9f48'
-
(internal is the context, 2004 is the extension of the SipTone)
The problem is that I do not hear the vm-login message. The PHONE has hung up
with a message 'Call Terminated' before that comes through.
The Grandstreams have no problem getting voicemail, and just to check out, I
disconnected the Siptone from the circuit, set up one of my Grandstreams with
the 2004 extension, and it worked ok.
I have no problem making internal/external calls with it. Its just Voicemail.
Relevant bits of config files
SIP
[2004]
type=friend
secret=2004
host=dynamic
mailbox=2004
dtmfmode=rfc2833
context=internal
VoiceMail
2004=>2004,2004
Extensions,conf
exten => 88,1,Wait(1)
exten => 88,2,VoiceMailMain
exten => 88,3,Hangup
Thanks
--
Clive
Email : [EMAIL PROTECTED]
Alt : [EMAIL PROTECTED]
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Alt : 01952 402032
SIP : [EMAIL PROTECTED]
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[Asterisk-Users] Broadvoice experience
I really don't find all this trashing of BV on this discussion very fair. 1) They are a big commercial service, great 'unlimited' packages, most credible competitor to Vonage. The in-state-unlimited is a great POTS-replacement deal. 2) Unlike Vonage, they play along with Asterisk and other soft clients at no extra charge, even fixing some retry-bugs in the asterisk SIP channel module. Both 'Olle' patches work fine for me. I run asterisk on a NATed LAN, connecting to numerous providers (sipgate, stanaphone, FWD, Broadvoice, voipjet, simpletelecom, gafachi, .) plus several * sites I run for my (non-telecom) employer around the world. They all have their pluses & minuses. BV's big minus to me has been their customer service, and their website. They recently really improved the website, no longer requiring the use of some satanic Java/activeX horror just to review your account. The customer service needs a major upgrade at BV. I had a problem related to a number change I requested. This number change triggered a (silent) change to the SIP secret on my account, which the service rep could not detect. After 3 days of calls and emails (Mostly ignored) I reached the manager of the operation, who was really nice and injected a new, working SIP secret in their system, the whole thing took 30 seconds. This comes down to further required work on their customer-handling backend stuff. Any account changes should trigger an email with the new setup parameters. These setup parameters should be accessible on the website! (as all of the above-named competitors do!). If this were done properly, I would have saved a lot of my time and aggravation, and would have consumed NO expensive human-time on their end. You have to call or write (I did better with calling) to get your password from customer service, and the answer you get may not be correct! I think this is the result of a (flawed) policy to keep most of the turnkey customers from playing with SIP hardware configurations. I was within a millimeter of dropping the service this week, but after persisting, I reached two really nice customer-service guys on the phone, who for now convinced me (not via marketing blah, but their helpful attitude) to hang on. On Sat, 20 Nov 2004, Kannaiyan Natesan wrote: A complete rubbish service in the whole world, which is spoiling the asterisk mailing list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Just getting started...
> Here's my current plan: > > Sounds like a plan? You asked for advice, here comes some that few will approve of :) FWIW I tried to get gnophone running and got no further than you did. What struck me though was that I have a very linux wise programmer friend and associate that never got it running either. The unpalatable advice of mine is that for initial testing, it would be good if you could either have a Windows box or laptop to avail yourself of the larger number of softphones, among them a few that work pretty well, or, bite the bullet and buy an IAXy if you can afford it and feel you'll be investing in one later. Or try SIP just to get things running. hth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug with Dial in AGI script?
Hi all,
I wrote an AGI script in perl i asked the
script to dial a number
$AGI->set_callerid($calleridnum);
if($left >
0) {
$AGI->exec('Dial',"H323/$phonenumber");
}else{ $res =
&mystreamfile("vm-goodbye");
$AGI->hangup(); }
The chan_h323 registers well to my gnugk. I call
the asterisk through the gatekeeper(GNUGK) using a CISCO ATA. The
prompt from Asterisk PBX until database manipulation is fine. But when
it dials the numbers, it rings but as soon as someone picks up the call
drop.
On Asterisk i allowed the following codec: g729,
gsm, ulaw and alaw. I purchased a g729 license from Digium. I know my
PSTN provider supports G729 codec and G711, and my cisco is set to the G729
codec.
Can someone give me a hint?
Thanks
Kido
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Re: [Asterisk-Users] * and NAT
> I'm having issues with * and getting connections outside of my local > lan . My * server is sitting behind a firewall which performs NAT . In the general section if sip.conf srvlookup = yes nat = yes won't hurt. qualify sometimes causes problems, I'd remove it until all is working, but when used should take a number as in qualify=300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Broadvoice
A complete rubbish service in the whole world, which is spoiling the
asterisk mailing list.
I have seen other companies also use asterisk, but they don't do this
gimmicks too for marketing using this mailing list.
Also I had some respect on Olle before, but I lost it now. (When people get
some money will they?)
Sending emails like Viruses (Exact format what virus emails will do). These
people never care for their customers.
See how we will feel if we get a message from a telephone service company.
"If you don't follow this, we will terminate your account" when it is not
our fault. Is it not a real nonsense? What they will do if we have a device
which sent similar messages to their server, do we need to upgrade our
firmware or to through out the device?
I too GRADE BROADVOICE AS A STUPIDIEST SERVICE I EVER SEEN in the broadband
voice services.
Cheap people will always give cheap prices, it is definitely true.
- Original Message -
From: "Ed Brady" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, November 19, 2004 10:01 PM
Subject: Re: [Asterisk-Users] Broadvoice
I also tried Broadvoice, I had the same problems and quality of service.
The real problems started however whenever I tried to cancel the
service
When I decided cancel, I contacted them 2 WEEKS before my billing date and
asked them to stop my account at the end of the month so as to not get
charged for another months worth of service. I would still have 2 weeks of
service left, however I had already switched my Asterisk box over to use
Nufone and Gafachi so it didn't matter,I also notified Broadvoice
using e-mail, which their support had instructed me to do, and I got a
reply indicating they had received and was sorry that I was leaving...
Well, they never cancelled the account, and two weeks later, I got an
email from Broadvoice saying my Credit Card had been billed for another
months service.
When I called to ask them for a refund and demand that the account be
cancelled, I had some bozo in support searching thru my call records to
make sure that I had not made a call on the account that THEY neglected to
close. The support guy then claimed that I had made a phone call on the
account and was not due the refund. Ok, stop for a minute; 1) How could I
have made the phone call if they would have closed the account when I had
instructed them too? 2) I hadn't made any phone calls with them for two
weeks! 3) Why was this guy searching my call records for a simple
account cancelation dispute?
I had to end up filing a complaint with my credit card company to get the
refund.
My suggestions:
Stay away..
Tim Mattison wrote:
Nah, tried it. I've been having these problems with BroadVoice from the
get-go. Randomly it'd go down for a few days without so much as a
simple apology. They'd suggest that I fiddle with the settings but
every time after a few days it'd just start working again (and it'd
actually work well when it was up).
I've given up on BroadVoice. The way they treat me shows me that they
simply don't want/need my business.
On Fri, 2004-11-19 at 11:24 -0600, Jay Milk wrote:
Use sip.broadvoice.com and if it doesn't resolve properly by itself, add
the sip.broadvoice.com as 147.135.8.128 to your etc/hosts file
-Original Message-
From: Tim Mattison [mailto:[EMAIL PROTECTED] Sent: Friday, November 19,
2004 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice
My BroadVoice account has been down for over a week with neither an
explanation nor a service credit. Our problems may be a little
different though because I don't remember what happened when I tried to
dial out. I know that I do get a "Request Timeout" error while trying
to register though.
On Fri, 2004-11-19 at 08:39 -0600, Tim Jackson wrote:
Anybody else having broadvoice problems?
-- Executing SetAccount("SIP/101-d03b", "LD") in new stack
-- Executing Dial("SIP/101-d03b",
"SIP/[EMAIL PROTECTED]") in
new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 408 "Request Timeout" back from
147.135.0.128
== No one is available to answer at this time
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
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--
Tim Mattison <[EMAIL PROTECTED]>
Mattison & Rosenthal Consulting Inc.
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Re: [Asterisk-Users] res_config / realtime?
On Thu, Nov 04, 2004 at 04:20:53PM -0600, Matthew Boehm spake thusly: > But yes, RealTime is very nice. It is still in development and there is > progress to bring it to as many apps as possible. You can get the RealTime > MySQL driver here: app_realtime seems a lot like res_data which is also realtime ie. not requiring a reload. Are they the same thing or is there some sort of duplication of effort going on here? I need to pick one of these to manage my own configs. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgpOeBxNOR7nu.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Error during installation
On Fri, 2004-11-19 at 21:36 +, Robert Kerry wrote: [snip] > res_crypto.c:27:25: openssl/ssl.h: No such file or directory > res_crypto.c:28:25: openssl/err.h: No such file or directory [snip] Obviously /usr/include/openssl/ssl.h and /usr/include/openssl/err.h can not be found. Fix it by installing the openssl development package and retry. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Controlling Asterisk from PHP?
Hi, On Thu, 2004-11-18 at 23:28, Michael Vogel wrote: > I just downloaded it. Now I only need to know, how to include it in > asterisk. The documention is ... hmm ... ;-) Ah, yes. But if you know PHP there is not much different from other classes. Just create a new phpagi object and start calling functions in there. However, please do contribute documentation if you think it helps :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAXy Configuration
Did that, and still no. I'm wondering if this just wont work with a windows dhcp server. On Sat, 20 Nov 2004 03:29:42 +0100, Wilson Pickett <[EMAIL PROTECTED]> wrote: > > > I can't seem to get this device to grab an ip from dhcp. We have a > > Look at the list of DHCP on the server, and if possible delete all > leases. Then try to power up the IAXy. > > My own IAXy suddenly stopped working today. In fact, it was due to the > coincidence that its internal ip had changed (DHCP) and it didn't > receive the provisioning when the asterisk server changed its address > due to being behind a NAT router. > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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] IAXy Configuration
Never Got that far. The device wont get an ip from dhcp for me to provision it. . . On Sat, 20 Nov 2004 03:29:03 +0100, BetaTeilchen <[EMAIL PROTECTED]> wrote: > Erik, can you please post your config-file ? > > Erik Espinoza schrieb: > > > > >I can't seem to get this device to grab an ip from dhcp. We have a > >working dhcp server (unfortunately it is on Windows), but I don't show > >any leases requested by the iaxy. > > > >Anyone have any ideas? > > > >The ethernet and phone lines are plugged in before the device is powered. > > > >Thanks, > >Erik > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >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 [EMAIL PROTECTED] 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] block caller id
Hi, > I have a PRI card. How do I block a caller id sent out to PSTN from a > SIP client? I add a remote-party-id field "privacy=full" but still get > caller id on a PSTN phone. I think that doing SetCIDNum() (with no args) before dial will do the trick. Matteo -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Starting AGI when handset is picked up?
Michael Vogel wrote: David Boyd schrieb: Please read wiki pages on the configuration files. This is prominently displayed for your perusal! Immediate=yes in zapata.conf will start the call immediately from the s extension! I think I have a problem in finding relevant information in this wiki ... Maybe since I don't use zapata I haven't found it. But I use a PhoneJack and in the phone.conf there is a similar option: "mode=immediate" It works. Now I have got to find out how to make AGI play the dialtone until a digit is entered. I found several commands like "Playtones" but it doesn't work ... Have a look at DISA: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA Excerpt: DISA (Direct Inward System Access) Allows someone from outside the telephone switch (PBX) to obtain an "internal" system dialtone and to place calls from it as if they were placing a call from within the switch. A user calls a number that connects to the DISA application and is given dialtone and context. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] block caller id
Title: block caller id Hi, I have a PRI card. How do I block a caller id sent out to PSTN from a SIP client? I add a remote-party-id field "privacy=full" but still get caller id on a PSTN phone. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Starting AGI when handset is picked up?
David Boyd schrieb: Please read wiki pages on the configuration files. This is prominently displayed for your perusal! Immediate=yes in zapata.conf will start the call immediately from the s extension! I think I have a problem in finding relevant information in this wiki ... Maybe since I don't use zapata I haven't found it. But I use a PhoneJack and in the phone.conf there is a similar option: "mode=immediate" It works. Now I have got to find out how to make AGI play the dialtone until a digit is entered. I found several commands like "Playtones" but it doesn't work ... Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and NAT
Hi All , I'm having issues with * and getting connections outside of my local lan . My * server is sitting behind a firewall which performs NAT . I have setup the port forward , configured externip= xxx.xxx.xxx.xxx and also have configured the client phones with nat=yes , careinvite=no , and qualify=yes . The error Im getting after it attempts a native bridge between my * server and a client over the internet is: "Warning[3197]: chan_sip.c:678 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1469" The error is giving me the client's internal IP address , though it is sitting behind another Nat'd internet connection. Connections on my Lan are working , Has anyone got any suggestions? Thanks in advance JK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
