Re: [Asterisk-Users] Making a Skinny phone talk to Asterisk
At 15:07 1-11-2003 -0600, you wrote: Last I checked skinny firmware would try to connect to a host that would resolve to CiscoCM1 Actually that is just a last-resort. Before that it will try and find the callmanager by looking for some special DHCP flag, and if that is not around it will try the setting for next-server (which is a DHCP option also). Some phones even try the DHCPserver itself if the above fails. I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
WipeOut wrote: David Sussman wrote: Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David Works fine on RH9.. I have a basic install guide.. http://members.lycos.co.uk/wipe_out/asterisk/ New page created on the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+linux+distributions Please add info on Linux distributions supported, both for Asterisk in general and zaptel device drivers. Also added this question to the FAQ. Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] NAT router and off-premise SIP audio problem
Rich, thank you for your informative reply. I checked with our admin and he replied: I setup from the start nat=yes and canreinvite=no on sip phones from Internet and modified the rtp channels (voice ports) and the rtp port on the phones. Still have the same problem, no sound. Perhaps the VPN solution is something we should try but this is more limiting than we had wanted... the concept that we could simply attach a SIP phone to a high speed internet connection anywhere, anytime (such as at a hotel when traveling) and become one with our office was a compelling one. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Saturday, November 01, 2003 8:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT router and off-premise SIP audio problem Jim, Off-premise SIPs are all behind simple NAT routers. Off-premise SIPs have been able to receive calls from and make calls through the PSTN. No problem. Calls between on-premise SIPs, not a problem. Calls between off-premise SIPs and any other SIPs connected to the server are a problem... they ring up but no audio is passed in either direction. SIP.CONF has NAT=YES. We presume that a dedicated IP address for the Asterisk server would resolve this but we would like to avoid the extra expense. What are we missing? TIA. It's the same problem that lots of others have posted about for months, and basically relates to not understanding the sip protocol during call setup. From a 10,000 foot view, here's what happens during call setup: 1. sip phone A dials sip phone B (communicates with * on udp 5060) 2. asterisk tells phone A to contact B directly (on udp 5060) and phone A does that (works since phone A is behind the nat box and is allowed the outbound dataflow) 3. phone A and phone B negotiate to establish the RTP channel (on some other udp port that is dependent upon the phone manufacturer) 4. phone A is allowed to communicate on that RTP port through the outbound nat box. 5. phone B is not allowed to pass inbound through the nat box on the choosen RTP port (since RTP is used for voice, it fails). That last step is the problem. You only have three choices today to fix the RTP problem in your case: 1. use the canreinvite=no statement on the phone definitions in sip.conf (which then forces all RTP sessions to pass through the asterisk box, increasing the processor workload of the box), or, 2. map each of the internal sip phones to a real registered IP address on the outside of the nat box. (Cheap nat boxes usually don't have this capability, however more expensive routers and firewalls do.) 3. replace the nat boxes with the VPN equivalents, and use the VPN tunneling to force the external phones to appear on the inside of your asterisk network. In those cases where there is only a single sip phone behind the nat box (and assuming a cheap nat box), one can change the RTP port range on some sip phones to some small specific set of udp ports, and then map those udp ports in the nat box to the individual internal sip phone. On the Cisco 7960 phones, the RTP port range can be set via Settings, SIP Config, item 16 (Start Media Port) and item 17 (End Media Port). One udp port will be required for each simultanous conversation supported by the sip phone, therefore on a six-line phone using a udp port range with at least six ports should work just fine. Also note that not all nat boxes work the same. Some vendors include special functions (and their marketing people exclude that technical detail in their published data), while others boxes are just plain dumb nat boxes. The only realistic way to see what is going on is to use a packet sniffer (like ethereal) to actually observe what the phone and nat box is really doing. Some working nat config's are just now beginning to get documented at the http://www.voip-info.org site. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Good system board to use with TE410P?
Hi- I'm looking for an appropriate system board to power a system with two (2) Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm considering vendors like Tyan for the motherboard. Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] recording files for menues
How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] Mobile: 972-55-229220
Re: [Asterisk-Users] recording files for menues
Shoval Tomer wrote: How do you suggest doing that? How can I convert wav files to gsm files? thanks #!/bin/sh for i in *.wav; do sox $i -r 8000 `basename $i *.wav`.gsm resample -ql; done ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good system board to use with TE410P?
Hi Scott, I use a Tyan 2665 (7505 based) M/B with a TE410P. That works well. This is a development workstation, so its probably not the kind of board you want for deployment. Regards, Steve Scott Stingel wrote: Hi- I'm looking for an appropriate system board to power a system with two (2) Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm considering vendors like Tyan for the motherboard. Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trustix support..
Has anyone tried running Asterisk under Trustix? (or Tawie as it is now called) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
Either it's not working, or I don't know what I'm doing. It's giving me the error sox: effect '.gsm' is no known! Let's say I need to convert file 1.wav to 1.gsm. How do I apply this command to it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel Betel Sent: Sunday, November 02, 2003 5:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] recording files for menues Shoval Tomer wrote: How do you suggest doing that? How can I convert wav files to gsm files? thanks #!/bin/sh for i in *.wav; do sox $i -r 8000 `basename $i *.wav`.gsm resample -ql; done ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Shoval Tom wrote: Either it's not working, or I don't know what I'm doing. It's giving me the error sox: effect '.gsm' is no known! Sounds like your copy of sox was not compiled with gsm enabled.. or you put a space between the ...wav`.gsm bit check with a single file like this: $ sox file.wav -r 8000 file.gsm resample -ql Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one way sound with x-lite (sip) -second attempt
Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The soundmeter on X-lite shows activity ... (not muted, correct device...) When pressing numbers while having these silent calls in x-lite is playing DTMFs at the PSTN phone side. sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to allow=all [1*phonenumber*] type=friend username=NAME secret=testpass auth=md5 nat=no host=dynamic reinvite=no canreinvite=no dtmfmode=inband callerid=Test *phonenumber* context=sip-phone-out Any suggestions ? Thanks, Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Good system board to use with TE410P?
Hi Steve- Yes, I was looking more for a less robust board (with integrated AGP) that would be more appropriate in a 2U rackmount for my customers - don't need firewire, SCSI, USB etc). I didn't really want to go to the Xeon at all, except that it seems that the 3.3v PCI requirement seems to push me there. I know that there's a new 5v version of the 410 coming, but can't wait for it due to customer requirements. Thanks and regards Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Sunday, November 02, 2003 2:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Good system board to use with TE410P? Hi Scott, I use a Tyan 2665 (7505 based) M/B with a TE410P. That works well. This is a development workstation, so its probably not the kind of board you want for deployment. Regards, Steve Scott Stingel wrote: Hi- I'm looking for an appropriate system board to power a system with two (2) Digium TE410P cards. Since these cards require the 3.3 volt PCI, I'm considering vendors like Tyan for the motherboard. Can anyone please tell me their experiences with the Tyan i7501 series (Xeon-basd), or recommend an alternate motherboard? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RH9 or RH8?
Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. Quoted from Paul Cheng, at 5 pm yesterday: I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. Quoted from Dustin Wildes, Wed 2003-10-29 10:07: All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. -- when discussing T1 card with voice and data transitting on it. Other than that, RH9 is fine. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quick Question
Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. There is no problem with RH9 and Zaptel drivers at this time. There might have been months ago when v9 first appeared on the scene, but all is well now. Ours have been running fine and stable for months with Zaptel. :) FWIW, each RH version from 7.0 to current has improved the video detection and support drivers (and thus X11 stuff) as have other linux distros. If the Netfinity 4000R can support 800x600 or better resolution, there is a high probability the v9 X11 stuff will work; might take some playing around substituting drivers though. We even have one new $350 Emachine with a Celeron (very cheap) and RH9 working just fine with asterisk, as well as a higher-end Dell 1-ghz laptop. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this flag? Or do you not use up2date to get and latest kernel and source? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this issue? Or do you not use up2date to get the latest kernel and source? On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsubscribe
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Re: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
To try and put it simply, the zaptel drivers will not compile with the -DZAPTEL_NETWORK flag (as set, not my default, in the Makefile), with any stock kernel including and after 2.4.21, which is when the new HDLC structure was imported from the development kernel tree. Therefore, it should be perfectly fine to run RedHat 9 or whatever as long as you installed (probably manually for RedHat) a stock kernel of version 2.4.20. Mind, however, that I do not have a RedHat box and that RedHat has historically made pretty extensive changes to a lot of the normal defaults to a lot of things, so the above statement may not necessarily be true. On Sunday, 02 November, 2003 11:22, Ray Burkholder wrote: All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this flag? Or do you not use up2date to get and latest kernel and source? Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Shoval Tom wrote: Either it's not working, or I don't know what I'm doing. It's giving me the error sox: effect '.gsm' is no known! Let's say I need to convert file 1.wav to 1.gsm. How do I apply this command to it? FAQ. See http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ (I've just added information on sound files. Thank you for the hint! :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way sound with x-lite (sip) -second attempt
Hi! reinvite=no canreinvite=no Don' these options have the same meaning? Just wondering... P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Threeway calling leaves outside trunks bridged
I think I found another interesting 'feature' with threeway calling. If you hang up while on a 3 way call with both parties on outside lines, Asterisk ends up removing the conference initiator and leaving the outside trunks bridged together. Is this a good idea? This could cause congestion problems on small configurations with limited outgoing lines. Maybe we should add an option to zapata.conf which forces 3 way calls to be completely dropped when the initiator hangs up on a conference with 2 outside lines bridged. Note: if one of the conference members is an internal extension, then this case should not not apply. Steve Rodgers San Diego CA -- Starting simple switch on 'Zap/3-1' -- Executing Dial(Zap/3-1, Zap/g1/9www8531212) in new stack -- Called g1/9www8531212 -- Zap/1-1 answered Zap/3-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Starting simple switch on 'Zap/3-2' -- Started three way call on channel 3 -- Started music on hold, class 'default', on Zap/1-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Executing Dial(Zap/3-2, Zap/g1/9www2891212) in new stack -- Called g1/9www2891212 -- Zap/2-1 answered Zap/3-2 -- Attempting native bridge of Zap/3-2 and Zap/2-1 -- Building conference on call on Zap/3-1 and Zap/3-2 -- Stopped music on hold on Zap/1-1 -- Attempting native bridge of Zap/3-1 and Zap/1-1 -- Attempting native bridge of Zap/3-2 and Zap/2-1 == Spawn extension (house-admin, 98531212, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' -- Hungup 'Zap/1-1MASQ' -- Attempting native bridge of Zap/1-1 and Zap/2-1 -- Why can't we hang up everybody in this case? -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
So true, yet so irrelevant for my purposes. I needed to convert existing IVR sound files to gsm, in order to demonstrate asterisk's functionality to my bosses (the ones who'll pay for the hardware, eventually...) Besides, even if I didn't have the files ready, I wouldn't use my lovely voice for it - I'll go to a recording studio with a professional (talking about a production environment) so it's good to know how to do this yourself, in case the studio doesn't know how to record them in this format. Thanks for the suggestion anyway. For you even more lazy ones - just leave yourself a message in asterisk's voice mail and go look for the file, it's there somewhere. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Sunday, November 02, 2003 8:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] recording files for menues Why do things the hard way? ; used to record prompts exten = 205,1,Wait(2) exten = 205,2,Record(/tmp/asterisk-recording:gsm) exten = 205,3,Wait(2) exten = 205,4,Playback(/tmp/asterisk-recording) exten = 205,5,Wait(2) exten = 205,6,Hangup bkw On Sun, 2 Nov 2003, Shoval Tomer wrote: How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] Mobile: 972-55-229220 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Any ideas? Btw, the first answer I got worked, I mistook ` for ' (newbie error, I know...) To be more specific for you newbies out there Create a file containing: copy below this line #!/bin/sh for i in *.wav; do sox $i `basename $i .wav`.gsm;done up to this line save it in your path, or in the directory containing the files you want to convert do a chmod +x filename (where filename is the name of your saved file) now you can run it while in the directory and it'll convert all *.wav files for you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Sunday, November 02, 2003 9:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] recording files for menues Shoval Tom wrote: Either it's not working, or I don't know what I'm doing. It's giving me the error sox: effect '.gsm' is no known! Let's say I need to convert file 1.wav to 1.gsm. How do I apply this command to it? FAQ. See http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ (I've just added information on sound files. Thank you for the hint! :-) /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New IAX software phone (for WIndows platform)
Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device; - GSM codec support; - advanced phonebook(search/add/replace/delete); - 12 memories with one click access (just click one of the 12 buttons to directly dial the number); - you can memorize IAX type addresses then call them with a click of a button; - 99 memories with two keys access (Mxx); - unlimited number of memory locations (just limited by your HDD capacity), accessible through the phonebook interface; - can dial directly from the phonebook; - can use separate audio device than the default one (you can play MP3's through your soundcard/speaker and use an USB headset fort phone purpose); - digital VU-meter (you can enable/disable it); - digital volume control (Vol UP / Vol Down); - redial/callwaiting callerID functionalities; - can switch between two calls; - out/in/missed/rejected/all calls list; - missed calls indicator; There is no help file available for the moment. I hope to finish it in a couple of days. Please send me your feedback. Thank you, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
Besides, even if I didn't have the files ready, I wouldn't use my lovely voice for it - I'll go to a recording studio with a professional (talking about a production environment) so it's good to know how to do this yourself, in case the studio doesn't know how to record them in this format. For professional recording you can use the same voice as the original prompts.. For details see http://www.digium.com/index.php?menu=thevoice The price seems reasonable to me.. According to John Todd's site the turnaround can be rather fast. (http://www.loligo.com/asterisk/sounds/Sounds-README.txt) http://www.loligo.com/asterisk/ for access to his directory of additional prompts. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Sorry... I was a bit in a hurry, and indeed I cannot expect all list readers to know about shell scripts... will elaborate a bit more in the future. I noticed you removed the sox resample -ql options, which on my studio recorded .wav files helped a bit, also It might be sensible to add a -c 1 to make sure sox will convert a stereo file to a single channel .gsm Regards, Michiel Shoval Tom wrote: Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Any ideas? Btw, the first answer I got worked, I mistook ` for ' (newbie error, I know...) To be more specific for you newbies out there Create a file containing: copy below this line #!/bin/sh for i in *.wav; do sox $i `basename $i .wav`.gsm;done up to this line save it in your path, or in the directory containing the files you want to convert do a chmod +x filename (where filename is the name of your saved file) now you can run it while in the directory and it'll convert all *.wav files for you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording files for menues
Shoval Tom wrote: Olle, I can't reach the faq page, and haven't been able to for the last four days. I'm getting 504 gateway timeout errors. Gateway timeout indicates something with your web proxy ...or? I've been able to reach the Wiki all weekend, I've updated and created several pages... I also now that Jim have been working to speed things up, among them adding more SQL connections as we have had many hits at the same time when mailing a URL to the list... You should be able to reach it. Is it only this web site or do you get that error message somewhere else? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a bit frightened, guys
Hi, I started looking into asterisk cause we're looking for a real-world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization to save on long distance and international calls. I've been playing with asterisk for a week now, and I am charmed (if not madly in love) with it. Today I went on and bragged to my boss about it and how we can implement it instead of buying something like Cisco's call manager. He got excited too and wants to have an estimate of hardware costs for a solution that'll work for us. Suddenly I was weak at the knees I have a couple of questions for you guys. Has someone tried this in a real world, production environment? What is the Asterisk server hardware recommendation for managing approx. 75 extensions and 16 analog lines? What telephony hardware do I need in order to get all these extensions and lines connected to Asterisk Do I need to replace our lines (analog) with a PRI line? Can I use the existing infrastructure (connecting the existing extensions to asterisk, for instance) What is there to be said for network bandwidth consumption for this size of a deployment? Do I need dedicated bandwidth for it to work properly Anything else I might have left out See, I'm not afraid of hard work, and I already started working on solutions not embedded into the Asterisk package (like broadening the web interface, having a directory application that connects to our internal phone directory, etc.). I am afraid that since this is not a commercially available product, since there's no guaranty it'll work I might find myself holding both ends of the short stick. For instance, can I get the hardware and return it after a while if it's not working? Where? This way, if it's a no go, I'm not stuck with useless hardware that cost thousands of dollars. In short, can Asterisk be deployed to a real production environment, like ours? Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] Mobile: 972-55-229220
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Dan, Looks great. Are you planning to release this with GPL? Peter At 22:21 2/11/03 +0200, you wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device; - GSM codec support; - advanced phonebook(search/add/replace/delete); - 12 memories with one click access (just click one of the 12 buttons to directly dial the number); - you can memorize IAX type addresses then call them with a click of a button; - 99 memories with two keys access (Mxx); - unlimited number of memory locations (just limited by your HDD capacity), accessible through the phonebook interface; - can dial directly from the phonebook; - can use separate audio device than the default one (you can play MP3's through your soundcard/speaker and use an USB headset fort phone purpose); - digital VU-meter (you can enable/disable it); - digital volume control (Vol UP / Vol Down); - redial/callwaiting callerID functionalities; - can switch between two calls; - out/in/missed/rejected/all calls list; - missed calls indicator; There is no help file available for the moment. I hope to finish it in a couple of days. Please send me your feedback. Thank you, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
How should I configure Asterisk to allow this soft-phone to register? Please provide an example -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists Sent: Sunday, November 02, 2003 11:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] recording files for menues Besides, even if I didn't have the files ready, I wouldn't use my lovely voice for it - I'll go to a recording studio with a professional (talking about a production environment) so it's good to know how to do this yourself, in case the studio doesn't know how to record them in this format. For professional recording you can use the same voice as the original prompts.. For details see http://www.digium.com/index.php?menu=thevoice The price seems reasonable to me.. According to John Todd's site the turnaround can be rather fast. (http://www.loligo.com/asterisk/sounds/Sounds-README.txt) http://www.loligo.com/asterisk/ for access to his directory of additional prompts. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] a bit frightened, guys
Hi, I believe the issues raised by this message are the same as mine, more on a commercialsense than for self use, but mostly the same. I've seen posts where real-life installations are mentioned, but not a reference to how Asterisk is working on production (and productive) environments. Any experiences would be very welcome I believe, not only on pure technical, but wider,sense. Thanks Shovalfor raising this issue, and to All for keeping this list so alive. Best Regards Jos L. Perez - Original Message - From: Shoval Tomer To: [EMAIL PROTECTED] Sent: Sunday, November 02, 2003 16:36 Subject: [Asterisk-Users] a bit frightened, guys Hi, I started looking into asterisk cause we're looking for a real-world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization to save on long distance and international calls. I've been playing with asterisk for a week now, and I am charmed (if not madly in love) with it. Today I went on and bragged to my boss about it and how we can implement it instead of buying something like Cisco's call manager. He got excited too and wants to have an estimate of hardware costs for a solution that'll work for us. Suddenly I was weak at the knees I have a couple of questions for you guys. Has someone tried this in a real world, production environment? What is the Asterisk server hardware recommendation for managing approx. 75 extensions and 16 analog lines? What telephony hardware do I need in order to get all these extensions and lines connected to Asterisk Do I need to replace our lines (analog) with a PRI line? Can I use the existing infrastructure (connecting the existing extensions to asterisk, for instance) What is there to be said for network bandwidth consumption for this size of a deployment? Do I need dedicated bandwidth for it to work properly Anything else I might have left out See, I'm not afraid of hard work, and I already started working on solutions not embedded into the Asterisk package (like broadening the web interface, having a directory application that connects to our internal phone directory, etc.). I am afraid that since this is not a commercially available product, since there's no guaranty it'll work I might find myself holding both ends of the short stick. For instance, can I get the hardware and return it after a while if it's not working? Where? This way, if it's a no go, I'm not stuck with useless hardware that cost thousands of dollars. In short, can Asterisk be deployed to a real production environment, like ours? Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: [EMAIL PROTECTED] Mobile: 972-55-229220 ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003
Re: [Asterisk-Users] Host unspecified ??
Hi, Here is wath happens: Asterisk*CLIsip debug SIP Debugging Enabled Asterisk*CLI Nothing happens when I use 'sip debug'. It seems that sip doesn't work. Wim - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 30, 2003 9:02 PM Subject: Re: [Asterisk-Users] Host unspecified ?? Hi Wim, Citeren Wim Venneman [EMAIL PROTECTED]: I changed the host to a fixed ip address (host1=192.168.10.12 and host2=192.168.10.13) now the ip address shows up in the 'host' field = ok. Try to call, no succes, nothing happens! What's wrong? That's a bit difficult to determine without more info. Could you enter the command 'sip debug', try calling with the phones and then copy what the console says ? Feel free to send off-list :-) -- Met vriendelijke groet, Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quick Question
I recall in the archives somewhere, and through someone's post earlier today, that there is some sort of problem with RH9 with Zaptel (hardware) drivers and that RH8 is preferred. Do you recall what kind of problem? The only problem I have is an annoying echo that I haven't yet gotten rid of. the only problems ive had with redhat 9 is the new thread model. it can be solved using: export LD_ASSUME_KERNEL=2.4.2 (i think) before you start asterisk, i do this in the asterisk init script so i dont forget. duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PHP Manager examples
Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
As the library is under LGPL (is not true?), I intend to keep this application as a freeware only... Yep its LGPL. Play with it and try to use all the features, which are very intuitive. Its a start but having to restart when you change registration isn't very intuitive. But its an excellent point to start. Good luck. Thanks, bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hello Steve, You are exactly right about the DDI line and thank you for clearing up the 600 ohm loop. Can you tell me other electrical details? It's just that all the PBXes (that I know) uses different cards for DDI lines and analog extension lines and since the CO normally or at least, expected to be much further away than an extension phone, I was wondering if there's a difference in the electrical requirment. thanks again, patrick Steve Underwood wrote: Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Manager examples
Kevin Bockman wrote: Anyone have any example scripts in PHP that connect to the manager? I started a PHP * Manager API, modeled on the Perl API, but haven't had a lot of time to work on it. I'll be happy to give you what I do have. -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clearing Queue Stats?
Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clearing Queue Stats?
--- Ken Godee [EMAIL PROTECTED] wrote: Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00? ___ A reload from the console does that.. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... actually theres been an opensource multiplatform iax soft phone on sourceforge for a while now: http://iaxclient.sourceforge.net/ duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Manager examples
--- Steve Sobol [EMAIL PROTECTED] wrote: I started a PHP * Manager API, modeled on the Perl API, but haven't had a lot of time to work on it. I'll be happy to give you what I do have. ___ Sure, I'd appreciate that. All I really need to start is to get it to login properly, as I stated. I think I can get the rest. Thanks, Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Live real extensions.conf samples?
It would be nice to see a real extensions.conf from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial 911 in thier dialplan. That's just something you don't want to forget! Not to mention that a business type extensions.conf needs to have several class of restrictions for different departments/people, most just have everything available to everyone, this is just not so in the real world. Not it mine anyway. If someone doesn't want to post you can alway email me direct. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
Hi, I started looking into asterisk cause we're looking for a real world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization - to save on long distance and international calls. I've been playing with asterisk for a week now, and I am charmed (if not madly in love) with it. Today I went on and bragged to my boss about it and how we can implement it instead of buying something like Cisco's call manager. He got excited too and wants to have an estimate of hardware costs for a solution that'll work for us. Suddenly I was weak at the knees I have a couple of questions for you guys. A. Has someone tried this in a real world, production environment? Yes, I have. It works fine if configured and managed properly. Only when you start to use the esoteric features do some ragged edges start to show, or if you plan on scaling to very large installations you will also need to dedicate more resources to the project (as would be the case in all such installations.) Additionally, if you pick hardware that is faulty or inappropriate for your environment, your users will have a bad experience despite what great things Asterisk can do. B. What is the Asterisk server hardware recommendation for managing approx. 75 extensions and 16 analog lines? This is not a clear question, though you may think it is. Your budget, your current configuration, your desire to move to VoIP phones, and your ease-of-use requirements all fit into this equation. Do you want VoIP phones on each desk? Do you want to keep all your existing analog phones? What features can you live without? Are all the analog lines in the same place? What is your bandwidth budget? What are you most common calling profiles? What does your long distance plan look like? [hint: don't answer all of these questions in a reply - it's much too long, and you should expect to find the answers in the history of the list if you read enough posts. If you are unable to answer these, then you should see my last comments in this reply.] C. What telephony hardware do I need in order to get all these extensions and lines connected to Asterisk See answer to B. D. Do I need to replace our lines (analog) with a PRI line? That would be optimal, yes, and would probably be cheaper for you in the long run. It seems that the most common breakeven point for PRI cutover is around 10-14 lines. E. Can I use the existing infrastructure (connecting the existing extensions to asterisk, for instance) Yes. See B. F. What is there to be said for network bandwidth consumption for this size of a deployment? Network between where and where? You haven't told us anything about what you intend on doing, and what the network would need to look like. G. Do I need dedicated bandwidth for it to work properly See F. H. Anything else I might have left out Quite a bit has been left out of your description. Do you want redundancy? What is your required uptime? Have you factored in a development platform for testing? What are your plans for long distance? What is driving this conversion, anyway? (price, features, flexibility, long-term costs, ???) See, I'm not afraid of hard work, and I already started working on solutions not embedded into the Asterisk package (like broadening the web interface, having a directory application that connects to our internal phone directory, etc.). I am afraid that since this is not a commercially available product, since there's no guaranty it'll work I might find myself holding both ends of the short stick. For instance, can I get the hardware and return it after a while if it's not working? Where? I don't know about this one. I would expect the answer is No unless the hardware is defective. This way, if it's a no go, I'm not stuck with useless hardware that cost thousands of dollars. Reward is not without risk. In short, can Asterisk be deployed to a real production environment, like ours? Yes. You would probably be wise to consult with someone who has had experience with building such systems before, so you avoid pitfalls and ask the right questions before you start buying gear. Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Mobile: 972-55-229220 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] a bit frightened, guys
Thanks for the detailed answer, and sorry about the not so detailed question. So here's my humble request. Can someone who has implemented a live production Asterisk deployment, preferably between two sites (HQ and a branch office, connected over the internet) spare the time and contact me here, or to my email directly? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Monday, November 03, 2003 1:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] a bit frightened, guys Hi, I started looking into asterisk cause we're looking for a real world solution. (when I say we I talk about a 50+ HQ and a 10+ branch office). We currently use a Panasonic analog PBX, with home-made IVR and PSTN lines. We'd like to deploy most of Asterisk's capabilities through out our organization - to save on long distance and international calls. I've been playing with asterisk for a week now, and I am charmed (if not madly in love) with it. Today I went on and bragged to my boss about it and how we can implement it instead of buying something like Cisco's call manager. He got excited too and wants to have an estimate of hardware costs for a solution that'll work for us. Suddenly I was weak at the knees I have a couple of questions for you guys. A. Has someone tried this in a real world, production environment? Yes, I have. It works fine if configured and managed properly. Only when you start to use the esoteric features do some ragged edges start to show, or if you plan on scaling to very large installations you will also need to dedicate more resources to the project (as would be the case in all such installations.) Additionally, if you pick hardware that is faulty or inappropriate for your environment, your users will have a bad experience despite what great things Asterisk can do. B. What is the Asterisk server hardware recommendation for managing approx. 75 extensions and 16 analog lines? This is not a clear question, though you may think it is. Your budget, your current configuration, your desire to move to VoIP phones, and your ease-of-use requirements all fit into this equation. Do you want VoIP phones on each desk? Do you want to keep all your existing analog phones? What features can you live without? Are all the analog lines in the same place? What is your bandwidth budget? What are you most common calling profiles? What does your long distance plan look like? [hint: don't answer all of these questions in a reply - it's much too long, and you should expect to find the answers in the history of the list if you read enough posts. If you are unable to answer these, then you should see my last comments in this reply.] C. What telephony hardware do I need in order to get all these extensions and lines connected to Asterisk See answer to B. D. Do I need to replace our lines (analog) with a PRI line? That would be optimal, yes, and would probably be cheaper for you in the long run. It seems that the most common breakeven point for PRI cutover is around 10-14 lines. E. Can I use the existing infrastructure (connecting the existing extensions to asterisk, for instance) Yes. See B. F. What is there to be said for network bandwidth consumption for this size of a deployment? Network between where and where? You haven't told us anything about what you intend on doing, and what the network would need to look like. G. Do I need dedicated bandwidth for it to work properly See F. H. Anything else I might have left out Quite a bit has been left out of your description. Do you want redundancy? What is your required uptime? Have you factored in a development platform for testing? What are your plans for long distance? What is driving this conversion, anyway? (price, features, flexibility, long-term costs, ???) See, I'm not afraid of hard work, and I already started working on solutions not embedded into the Asterisk package (like broadening the web interface, having a directory application that connects to our internal phone directory, etc.). I am afraid that since this is not a commercially available product, since there's no guaranty it'll work I might find myself holding both ends of the short stick. For instance, can I get the hardware and return it after a while if it's not working? Where? I don't know about this one. I would expect the answer is No unless the hardware is defective. This way, if it's a no go, I'm not stuck with useless hardware that cost thousands of dollars. Reward is not without risk. In short, can Asterisk be deployed to a real production environment, like ours? Yes. You would probably be wise to consult with someone who has had experience with building such systems before, so you avoid pitfalls and ask the right questions before you start buying gear. Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email:
Re: [Asterisk-Users] XTEN-Lite Bad sound!
WipeOut wrote: Jason A. Pattie wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ariel Batista wrote: | Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I have exactly this same problem as well. It's even worse when running X-Lite under Wine under Linux. Has the bad quality started just recently? Has it ever worked nicely for you?? If either of these is yes.. What has changed in your setup? Have you recently upgraded to a newer CVS?? I don't have an answer for you but at least it may stop others falling into the same problem if somthing can be identified as the cause.. later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users There was a thread a few days ago about changing the value on xlite of the silence suppresion, altough this guy reported bad/choppy sound with music on hold... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a bit frightened, guys
So here's my humble request. Can someone who has implemented a live production Asterisk deployment, preferably between two sites (HQ and a branch office, connected over the internet) spare the time and contact me here, or to my email directly? As a lurker, I would very much appreciate if this conversation could be kept on-list. Not only does it help more than just yourself then, but it also gets to be part of the archive which search engines can access. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Interesting thought, with these DDI lines a UK based company could easily get a good number of incoming analogue lines into an Asterisk system because teh FXS cards have far more ports than the FXO ones. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] surpress dial tone on TDM400p
i've already tried to change the indications.conf to the following:
dial = 0/1500
but the dial tone still persists
i am using the following workaround but obviously not a clean
b'cos it just replace dial tone with some other tone.
in zapata.conf
context=spec
immediate=yes
signalling=fxo_ks
channel=2 ; TDM400p-1
in extensions.conf
[spec]
exten = s,1,Background(pbx-silence)
exten = _,1,Dial(Zap/1/${EXTEN})
please help,
patrick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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[Asterisk-Users] Asterisk behind LinkSys NAT Routing
Problem I have is this. outside firewall
(extension 2003) can call me inside firewall (extension 2000) and all is
fine. If I call from inside firewall (extension 2000) to outside firewall
(extension 2003) I hear no ringing and person at other end can pick up and I
hear for maybe a half second then I go to voicemail. If I add another
extension on the outside then communication between outside and outside through
* is not possible at all. I know I can not be the only one who has tried
to do this. Please any help would be greatly appreciated.
My configuration.
Asterisk Server
--
Linux RedHat 9.0
Asterisk CVS Update - 11/02/03 around 10:00AM
PT
Zaptel CVS Update - 11/02/03 around 10:00AM
PT
LinkSys Router with Asterisk server set as DMZ
host.
Desktop Computer #1
---
Windows XP
Xten X-Pro build 1082
Behind same LinkSys router.
extension 2000 in asterisk
Desktop Computer #2
---
Windows XP
Xten X-Pro build 1082
Not behind any firewall.
extension 2003 in asterisk
sip.conf
; General definitions for the sip.conf
file.[general]port = 5060bindaddr = 0.0.0.0allow =
gsmcontext = bogon-calls ; Default any unknown
calls;[2000]type=friendusername=2000secret=grinchhost=dynamicdefaultip=192.168.1.210context=trustednat=yesqualify=1000mailbox=2000
;
[2003]type=friendusername=2003secret=grinchhost=dynamiccontext=normalnat=yescanreinvite=nomailbox=2003
extensions.conf
[globals]
; Variables to VoIP extensions by
nameROBERT=SIP/2000
[general]static=yes ; These two
lines prevent the command-line interfacewriteprotect=yes ; from overwriting
the config file. Leave them here.
[bogon-calls]; Bogus calls if they find there
way in to the system without authorization some how.exten =
_.,1,Congestion ; if someone accidentally finds there way here give them a fast
busy.
[stations]exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)exten
=
2000,103,Hangup
;exten = 2003,1,Dial(SIP/2003,20) exten
= 2003,2,Voicemail(u2003) exten = 2003,102,Voicemail(b2003)exten
=
2003,103,Hangup
;exten = 2997,1,VoicemailMain(2997) exten =
2998,1,VoicemailMain(2998) exten = 2999,1,VoicemailMain(${CALLERIDNUM})
;; Direct Dial. For those trusted to use thephone
properly.[directdial]exten = 9,1,Dial(Zap/g1/${EXTEN:1})exten
= 9,2,Congestioninclude = international;; International
calling code and prefix used for users trusted to make international
calls.[international]exten =
_9011.,1,Dial(Zap/g1/${EXTEN:1})exten = _9011.,2,Congestioninclude
= longdistance;; Long distance calling code and prefix used for
users trusted to make long distance calls.[longdistance]exten =
_91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})exten =
_91NXXNXX,2,Congestioninclude = local;; Local calling code
and prefix used for users trusted to make local calls.[local]exten =
_9NXX,1,Dial(Zap/g1/${EXTEN:1})exten =
_9NXX,2,Congestion;; Trusted users from sip.conf who are able to
fully use the phone.[trusted]include = stationsinclude =
directdial;; Normal users from sip.conf who are able to make local calls
only.[normal]include = stationsinclude = local;;
Public area for people who are only allowed to make calls to other
extensions[public]include = stations;; When someone calls
the work line of XXX-XXX- they are directed through
this.[inbound-work]exten =
s,1,Zapateller(answer|nocallerid)exten = s,2,Dial(${ROBERT},20)exten
= s,3,Voicemail(u2997)exten = s,4,Hangupexten =
s,103,Voicemail(b2997)exten = s,104,Hangup;; When someone calls
the home line of XXX-XXX- they are directed through
this.[inbound-home]
exten =
s,1,Dial(${ROBERT},20)
exten = s,2,Voicemail(u2998)exten = s,3,Hangupexten =
s,102,Voicemail(b2998)exten = s,103,Hangup
Robert
RE: [Asterisk-Users] New IAX software phone (for WIndows platform)
Any thoughts or plans on making it available on the asterisk key *NIX? AJ On Sun, 2 Nov 2003, Senad Jordanovic wrote: Finaly, someone has started the IAX soft phone ball :) Thanks, Dan... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
As the library is under LGPL (is not true?), I intend to keep this application as a freeware only... I want to add new features, but for one of them I need new functions implemented in the library (like multiple codecs support, message waiting indicator, conferencing, etc.). There is no requirement that you GPL or LGPL your code (other than the requirements that you publish changes to iax-client and/or libiax. However, if you elect to GPL your software, you can get assistance from other people around the net. In addition, since this is such an important project, I'm willing to personally any assistance you may need with regards to IAX/IAX2 if you're going to GPL or LGPL your final product. As a side note, I strongly would like to see someone implement a client using libiax2 which implements IAX2 instead of the (now obsolescent) IAX version 1. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommended places for beginner to start?
(The list may get this msg twice; I originally sent it from the wrong email address, my apologies. Moderator, if you can, please delete my original email submission from [EMAIL PROTECTED] Thanks.) Hello- Summary: Can anyone recommend a place to start to learn how to create an Asterisk system given a basic Digium PCI card and some generic PC hardware? Details: I plan to help a friend not familiar with Linux platforms load and evaluate a Digium/Asterisk system for business-development purposes. A couple years ago I used to work as a Unix/Linux sw developer and sysadmin, but have been doing sales/marketing stuff since. Where should I start to read about loading a system? My friend apparently has a $100-flavor of Digium for eval purposes (can hook up to one external phone line, or so I'm told), but knows little else. Since I've been the unix/linux geek in a past life, he came to me for assistance. I downloaded the .pdf handbook, and their appeared to be a reference to a downloading and installing section, but I couldn't find any text/body that actual described this process. Do I pick any linux flavor (presumably with compatible kernel) like RedHat/Debian/SuSE and load up the source/pkgs/rpms necessary and let 'er rip? Will I get a phone switch/PBX (or whatever this is) going fairly easily, assuming I get my linux box/platform fired up ok? Any gotchyas, tricks of the trade, things to know/worry about, etc? Is this all contained in the .pdf handbook? When I skimmed it, I didn't find anything that seemed to match up with a installing for a rookie's perspective like mine, but maybe I overlooked something. I have yet to get my hands on the Digium hardware/docs/etc that my buddy ordered; maybe some answers/secrets/support-resources are in there? I'm on vacation right now and am a little short on info, but before delving into this when I get back (probably starting around 11/5) I thought I would send out this note to the user list so that I might potentially save some time in research/pain before I start. Thanks for any help! -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hi Patrick, From memory (I haven't lived in the UK for 11 years) the electrical characteristics are pretty much the same: -48V, 35mA loop current, 600ohm complex impedance. One key difference is an extension needs to ring, but a DDI line does not. The different cards you see used may be because the DDI port has no ringer, or it may be a marketecture issue - they can probably squeeze more money out of the customers for the DDI ports. If they get full approvals only on the DDI card, then only a pricy DDI card can be used to attach to the DDI lines. Regards, Steve hkirrc.patrick wrote: Hello Steve, You are exactly right about the DDI line and thank you for clearing up the 600 ohm loop. Can you tell me other electrical details? It's just that all the PBXes (that I know) uses different cards for DDI lines and analog extension lines and since the CO normally or at least, expected to be much further away than an extension phone, I was wondering if there's a difference in the electrical requirment. thanks again, patrick Steve Underwood wrote: Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Questions
hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Read error on sound device
Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1217602880]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! When I start asterisk (./asterisk -c ) I can see following warning. [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[1167272000]: File chan_oss.c, Line 238 (sound_thread): Read error on sou nd device: Resource temporarily unavailable I was wondering that asteirsk couldn't find the sound card, because of this. However, I can use command 'dial' from CLI to dial out an extension and have a conversation using a headphone/mic. Also, I learned from various posts in the mailing list and found that I have to modprobe ztdummy, so I installed the Zaptel and ran modprobe before executing ./ asterisk. Even then the problem still exists. I really appreciate any help in fixing this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Manager examples
Here is my example. I'm using a lot of times a day. ?php $socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: admin\r\n); fputs($socket, Secret: blabla\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); $wrets=fgets($socket,128); ? Regards, Gus - Original Message - From: Kevin Bockman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 02, 2003 6:42 PM Subject: [Asterisk-Users] PHP Manager examples Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] msn messenger
On Sat, Nov 01, 2003 at 09:35:26AM +0100, Florian Overkamp wrote: At 01:43 1-11-2003 +0300, you wrote: Is msn messenger capable of using asterisk as it's gateway? Yes, provided you are using MSN 4.7, and not 5.0 or higher. Configure the Communications Service under the Options/Accounts pane. I'm not sure either way with MSN Messenger, but Windows Messenger (slightly different - same servers - no adds) which comes with XP does, there is a registry key (HKEY_CURRENT_USER-software-messenger_service-corpPC2phone) which you need to change to '1' to get the make a phone call link down the bottom of the window. Maybe there is a Wiki article about setting up softphones? cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Questions
I built something very similar using: - Adtran TA750 bought off Ebay for around $400 (you can do much better, I was in a hurry.) - A Digium Wildcard T100P - A 4 port FXO card for the TA750 (I searched Google for Adtran FXO and clicked one of the sposored links.) You might have to pick up some other misc bits pieces depending on what the Channel bank you get off EBay has. I had to buy a 25 pair cable with an Amphenol connector and a Type 66 punch down block. Right now my set up has an evil hum on outgoing calls. I suspect the home brew T1 reverse cable I'm using. HTH. brez wrote: hello, I am completely new to things but was wondering if some one could steer me in the right direction [i.e. i was volunteered to get a PBX running with little or knowledge] good news is, i got a lot of experience with open source / linux / etc. anyhow. we have 4 lines coming in and need 16 extensions. we have the PC and the 16 analog phones. the question is what type of hardware will i need? i.e. modem, a phone 'hub' [or whatever it is called for pluggin all the phone lines into] - basically a small office environment. if any of you using asterisk in a similar environment could spell out exactly what hardware youre using [and perhaps where to buy it] for your office, i would really appreciate the help. thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] which TDM to use? DID line from telco with no dial tone and no voltage
Hi again, thanks a million for the info. regards, patrick Steve Underwood wrote: Hi Patrick, From memory (I haven't lived in the UK for 11 years) the electrical characteristics are pretty much the same: -48V, 35mA loop current, 600ohm complex impedance. One key difference is an extension needs to ring, but a DDI line does not. The different cards you see used may be because the DDI port has no ringer, or it may be a marketecture issue - they can probably squeeze more money out of the customers for the DDI ports. If they get full approvals only on the DDI card, then only a pricy DDI card can be used to attach to the DDI lines. Regards, Steve hkirrc.patrick wrote: Hello Steve, You are exactly right about the DDI line and thank you for clearing up the 600 ohm loop. Can you tell me other electrical details? It's just that all the PBXes (that I know) uses different cards for DDI lines and analog extension lines and since the CO normally or at least, expected to be much further away than an extension phone, I was wondering if there's a difference in the electrical requirment. thanks again, patrick Steve Underwood wrote: Hi Patrick, You are in the UK, right (at least DDI strongly suggests that)? This is the commonest signalling for a DDI line on an analogue pair. The line is behaving just like the main exchange is a telephone. It picks up the line, by applying a 600ohm loop, and dials (with pulses per second or DTMF) into your PBX. Your PBX port is behaving like it is a public exchange, with a phone attached. Electrically, Digium's FXS card should do the job you need, but others will have to tell you whether * has the software features needed to make this work (it should certainly be pretty close). Regards, Steve hkirrc.patrick wrote: as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i disconnected a running DID line from our PBX and did a bunch of tests on it and found the following: * line from telco has NO voltage * the port from pbx is supplying the power(voltage) but no dial tone *the moment i disconnected the DID line from the PBX port, an alarm is triggered at the telco CO * i can attach an ordinary analog phone to the PBX PORT, pick up the handset and send (dial) 4 dtmf digits (being the last 4 digits of our DID number), the PBX will bridge me to the appropriate extension phone. * if or when the extension phone picks up, the PBX reverses the polarity on the line what type of signaling should i be using for such a line? many thanks in advance, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended places for beginner to start?
On Sun, Nov 02, 2003 at 05:56:02PM -0700, Matthew England wrote: (The list may get this msg twice; I originally sent it from the wrong email address, my apologies. Moderator, if you can, please delete my original email submission from [EMAIL PROTECTED] Thanks.) Hello- Summary: Can anyone recommend a place to start to learn how to create an Asterisk system given a basic Digium PCI card and some generic PC hardware? I started with: http://www.automated.it/guidetoasterisk.htm probably low enough level for someone coming from a Windows admin background, perfect for someone who has been in Marketing Land (anyone read Dilbert? :-) for a while. Enough to get up a demo system. cheers, Woody Details: I plan to help a friend not familiar with Linux platforms load and evaluate a Digium/Asterisk system for business-development purposes. A couple years ago I used to work as a Unix/Linux sw developer and sysadmin, but have been doing sales/marketing stuff since. Where should I start to read about loading a system? My friend apparently has a $100-flavor of Digium for eval purposes (can hook up to one external phone line, or so I'm told), but knows little else. Since I've been the unix/linux geek in a past life, he came to me for assistance. I downloaded the .pdf handbook, and their appeared to be a reference to a downloading and installing section, but I couldn't find any text/body that actual described this process. Do I pick any linux flavor (presumably with compatible kernel) like RedHat/ Debian/SuSE and load up the source/pkgs/rpms necessary and let 'er rip? Will I get a phone switch/PBX (or whatever this is) going fairly easily, assuming I get my linux box/platform fired up ok? Any gotchyas, tricks of the trade, things to know/worry about, etc? Is this all contained in the .pdf handbook? When I skimmed it, I didn't find anything that seemed to match up with a installing for a rookie's perspective like mine, but maybe I overlooked something. I have yet to get my hands on the Digium hardware/docs/etc that my buddy ordered; maybe some answers/secrets/support-resources are in there? I'm on vacation right now and am a little short on info, but before delving into this when I get back (probably starting around 11/5) I thought I would send out this note to the user list so that I might potentially save some time in research/pain before I start. Thanks for any help! -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * troubles
Hello all, Been a while since I've strolled this way.Apologies in advance if this is a common line of questioning. I've just bought a new Intel 865G based board with a P4 Hyperthreading processor. I believe I've gottenSMP set up correctly: in the menuconfig I specified SMP and told acpi to enumerate processors. Did I leave out anything? Anyway, the dmesg looks good and the server doesn't freeze or blow up. In the zaptel makefile, I uncommented the flag for SMP. My problem is that on my TDM20B, I lose dialtone after a while. One time I lost dialtone but had battery, another time lost both dt and battery. A reboot brings it back. Suggestions? TIA, Victor
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Dan. thanks for good application! and I wish 'no with installer' package about that. because I think use with USB-memory device in any places (ie.net-cafe.) is that need registry setting or not? On Sun, 2 Nov 2003 22:21:09 +0200 Dan [EMAIL PROTECTED] wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Robert,
Try adding canreinvite=no to extn 2000 and reload asterisk. In your
specific case, it needs to be on each sip.conf extn definition.
Rich
Problem I have is this. outside firewall (extension 2003) can call me inside
firewall (extension 2000) and all is fine. If I call from
inside firewall (extension 2000) to outside firewall (extension 2003) I hear no
ringing and person at other end can pick up and I
hear for maybe a half second then I go to voicemail. If I add another extension on
the outside then communication between
outside and outside through * is not possible at all. I know I can not be the only
one who has tried to do this. Please any help
would be greatly appreciated.
My configuration.
Asterisk Server
--
Linux RedHat 9.0
Asterisk CVS Update - 11/02/03 around 10:00AM PT
Zaptel CVS Update - 11/02/03 around 10:00AM PT
LinkSys Router with Asterisk server set as DMZ host.
Desktop Computer #1
---
Windows XP
Xten X-Pro build 1082
Behind same LinkSys router.
extension 2000 in asterisk
Desktop Computer #2
---
Windows XP
Xten X-Pro build 1082
Not behind any firewall.
extension 2003 in asterisk
sip.conf
; General definitions for the sip.conf file.
[general]
port = 5060
bindaddr = 0.0.0.0
allow = gsm
context = bogon-calls ; Default any unknown calls
;
[2000]
type=friend
username=2000
secret=grinch
host=dynamic
defaultip=192.168.1.210
context=trusted
nat=yes
qualify=1000
mailbox=2000
;
[2003]
type=friend
username=2003
secret=grinch
host=dynamic
context=normal
nat=yes
canreinvite=no
mailbox=2003
extensions.conf
[globals]
; Variables to VoIP extensions by name
ROBERT=SIP/2000
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
[bogon-calls]
; Bogus calls if they find there way in to the system without authorization some
how.
exten = _.,1,Congestion ; if someone accidentally finds there way here give them a
fast busy.
[stations]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
;
exten = 2003,1,Dial(SIP/2003,20)
exten = 2003,2,Voicemail(u2003)
exten = 2003,102,Voicemail(b2003)
exten = 2003,103,Hangup
;
exten = 2997,1,VoicemailMain(2997)
exten = 2998,1,VoicemailMain(2998)
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
;
; Direct Dial. For those trusted to use the phone properly.
[directdial]
exten = 9,1,Dial(Zap/g1/${EXTEN:1})
exten = 9,2,Congestion
include = international
;
; International calling code and prefix used for users trusted to make international
calls.
[international]
exten = _9011.,1,Dial(Zap/g1/${EXTEN:1})
exten = _9011.,2,Congestion
include = longdistance
;
; Long distance calling code and prefix used for users trusted to make long distance
calls.
[longdistance]
exten = _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _91NXXNXX,2,Congestion
include = local
;
; Local calling code and prefix used for users trusted to make local calls.
[local]
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,Congestion
;
; Trusted users from sip.conf who are able to fully use the phone.
[trusted]
include = stations
include = directdial
;
; Normal users from sip.conf who are able to make local calls only.
[normal]
include = stations
include = local
;
; Public area for people who are only allowed to make calls to other extensions
[public]
include = stations
;
; When someone calls the work line of XXX-XXX- they are directed through this.
[inbound-work]
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,Dial(${ROBERT},20)
exten = s,3,Voicemail(u2997)
exten = s,4,Hangup
exten = s,103,Voicemail(b2997)
exten = s,104,Hangup
;
; When someone calls the home line of XXX-XXX- they are directed through this.
[inbound-home]
exten = s,1,Dial(${ROBERT},20)
exten = s,2,Voicemail(u2998)
exten = s,3,Hangup
exten = s,102,Voicemail(b2998)
exten = s,103,Hangup
Robert
---End of Original Message-
___
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[Asterisk-Users] Questions from a total beginner
Hello, I would like to setup an * system but have no experience with Linux and am just learning about VoIP. My programming experience is pretty limited as well, so I may be getting in way over my head, but I am willing to take the time to figure out how to use *. I'd like to use * to create a PBX that initially would include myself and threeor four individuals who are located inseparate locations around the US. After getting that to work, I'd also like to set up * at another location to provide a local PBX for a small office environment (just 3 or 4 extentions), that would be linked with the * server at my location.It's my understandingthat would enable communication via the net to and from my location and the city where the second * server would be located, thus eliminating any long distance charges for calls between the two locations. I have twoPC'sthat I want to network together usingLinksys 802.11g gear (WRT54Gap/router aWMP54G PCIcard in my *server). My main machine isan XP. The one I am planning to usefor the * serverhas an AMD 500 processor; 64mb ram; and 30+ gb of hard drive available.I've downloaded the RH9 iso files to install Linux on the proposed server.I also have one phone line coming into my home that I would connect to the * server with a Wildcard X100P. Hopefully I've providedenough backgroundinfo that my questions will make sense. 1)From what I've read, the hardware for my proposed * server is adequate. Is that correct?Should I put another stick of 64mb ram in the box? 2)Is there anything special I need to know about installing RH9 to work with * and what type of install is recommended? Also, it's my understanding that I'll have to install some additional drivers to getRH9 to work with the WMP54G PCI card,and maybe the WRT54G also. I'm confused on that issue so any clarification would be appreciated. 3) I'd like to set up a VoIP phone at my location, but don't know what brand to use, nor the factors to consider in making that selection, so suggestions would be great. 4) Do the individuals at the otherlocations only need to obtain a VoIP phone and the appropriate sound card in order to gain access * at my location? Or is there some additional hardware/software required on their end of the connection? I assume that using the same VoIP phone at each location would be the ideal and I believe that's something we can do, if recommended. 5) My * server will be operating behindNAT on the broadband router, but from what I've read, that can work, although SIP phones can have some difficulty with NAT.Can the VoIP phoneused eliminateany problems with NAT? 6)What are the pros and cons if we were tohave thevarious locations (individuals and eventually the second * server)communicateover aVPN? Thanks in advance for any and all assistance. Roger
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Mark, There is no requirement that you GPL or LGPL your code (other than the requirements that you publish changes to iax-client and/or libiax. There is no change in the libiax for the moment. My DLL is just used to export the functions from the library to the main application. However, if you elect to GPL your software, you can get assistance from other people around the net. In addition, since this is such an important project, I'm willing to personally any assistance you may need with regards to IAX/IAX2 if you're going to GPL or LGPL your final product. I know that and I will think about it As a side note, I strongly would like to see someone implement a client using libiax2 which implements IAX2 instead of the (now obsolescent) IAX version 1. I really want to do it. They are a lot of things to be added in the library if I want to increase the low level functionality of my application. There is a libiax2 library for Windows available somewhere or I must build it myself based on libiax?? Thanks a lot, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
