Re: [Asterisk-Users] A new alternative to see who is online
Thanks for all your comment. I will release the source code but not for the moment. I really need to clean the graphical part but right away I don't have enough time. Regards, Nicolas Bruxer Lyle Giese wrote: 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 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] hangup()???
Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help ___ 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] H323 Problems
Peter, Peter Landy wrote: New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Did you apply the OpenH323 patch BEFORE configuring/compiling the library? Regards Peter Landy 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] Phones
Hello, On Sun, 21 Nov 2004, Tracy R Reed wrote: On Sun, Nov 21, 2004 at 04:25:39PM -0800, Tony Vickers spake thusly: What VOIP Phones is everyone using and why? Is the a common phone that seems the work the best? Just wondering. Snom - Good business class phones. Do they have any remote management functionality other than the web interface, something like tftp like the polycom and cisco? Been a while since I looked at them. Yes, they have. You can save the config on a http server and point the phone via dhcp to the right configuration file. You could even write a small webapp that generates the phoneconfig out of a database. Snom has a PDF on their website documenting these features. Look for mass deployment on their website. Torsten Krueger -- Media Online Internet Services Marketing GmbH Torsten Krueger [EMAIL PROTECTED] fon: 49-231-5575100fax: 49-231-55751098 Kurze Str. 10 D-44137 Dortmund ___ 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 error tolerence??
Hi, Didn't get any opinions on the log file I mailed onto the list over the weekend so I am continuing to try and track the cause for the dropped calls.. I have a feeling that its to do with IAX being way too sensitive when it comes to packet loss.. Since it is going across the internet it needs to be more tolerant when it comes to errors and packet loss.. Are there any settings (in the conf file or the source) that I can change to make it more able to keep the call connected without dropping it when there are errors even if the sound breaks up for a couple of seconds?? Later.. ___ 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] Phones
Tony Vickers wrote: What VOIP Phones is everyone using and why? Is the a common phone that seems the work the best? Just wondering. I am using X-Lite and Siemens Optipoint 400s, simply because there is a surplus of them at work from another installation. I've not had any trouble with them so far, but I have not yet found the information needed for distinctive ring. ___ 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 and H.323 Gatekeeper
I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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
Re: [Asterisk-Users] Gatway with IAX ?
Joseph wrote: If I want to use IAX instead of SIP, do I need to get gateway that support IAX. Are there such gateways? I plan to connect 3 to 4 standard phones via gateway with * In addition I don't want to use SIP to setup VoIP. IAX is more suitable for communication over firewall. Joseph perhaps you can have your clients connect using IAX, and your Asterisk box connect to a SIP server? Right now there aren't that many providers offering IAX support. SIP is by far the more popular, especially with voip hardware vendors. ___ 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 not correctly unregistered users...
Hi there, Last week we discovered some problems using NAT/Routers and SIP Clients. We had some work on our local network (exchanging routers) and so our clients were temporarily offline without unregistering correctly. In our mysql sipfriends table were several entries with the same IP (this is normal: our router) and different ports, but some clients (who were off/online) were in that database with the same ports. The Client, which later registered could not call, because chan_sip.c failed to authenticate the user (two entries in the database with the same IP / port). Is there a workaround for this? Can i simply remove duplicate entries like this on registration? I could update the chan_sip.c myself, but i am wondering if someone else had this problem before. And before i alter the source, i would like to know, if this is a good idea to update the data-sets for the users, who have the same IP/port. If there are no solutions/experiences on this list, i will send my message to the -dev list as well. Thanks in advance, Carsten Bock ___ 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] E100P - Generic (Clone) - :)
Em Ter 09 Nov 2004 19:20, Richard Moore escreveu: Hello list , I´m happy to announce our first stable production of a E100P generic boards. The prices are : 50 units = US$ 112,50/unit 50 units = US$ 98,10/unit optional resources: consulting : 32 US$ / server - You need provide root access if needed. ( Only Linux supported ) Freight : FOB TAIWAN We accept VISA and Mastercard. Feel free to contact us. Kind Regards, Richard Moore Asterisk Senior Consultant / Engineer Taiwan Was this a hoax ? I e-mailed him twice over 2 weeks, no answer... Anybody able to contact this guy ? Marcelo Pacheco ___ 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] H323 Problems
Peter, If you have the lastest CVS version of asterisk(1.0.11) , and the latest version of asterisk-oh323(0.7.0), it won't work. What version of asterisk are you running? what version of oh323 are you trying to compile? K. Peter, Peter Landy wrote: New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Did you apply the OpenH323 patch BEFORE configuring/compiling the library? Regards Peter Landy 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 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 and H.323 Gatekeeper
Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way. K. - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 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] H323 Problems
Yes I did. Does anyone have a working list of libraries and versions. I have tried with different releases of H323 and they all give different errors. Also is it necessary to compile the H323 under asterisk src/channels/H323 as this also bails on errors. The rest of my asterisk install works perfectly it is just H323 I am hung up on. Cheers Pete -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: 22 November 2004 09:01 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 Problems Peter, Peter Landy wrote: New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Did you apply the OpenH323 patch BEFORE configuring/compiling the library? Regards Peter Landy 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] Asterisk and H.323 Gatekeeper
Thank you, I will need a SIP client with G723 and/or G.729 then. Do you know any sip clients that do both ? Regards, Jorge A. -Mensaje original- De: kido noagbodji [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 8:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way. K. - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 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] A new alternative to see who is online
Great program, thanks! only one question: when i reboot my pc i can't see who is online, until the sip user re-register their clients at the server. leon only seems to update his online status when a sip client connects. Jens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Sent: Monday, November 22, 2004 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A new alternative to see who is online Thanks for all your comment. I will release the source code but not for the moment. I really need to clean the graphical part but right away I don't have enough time. Regards, Nicolas Bruxer Lyle Giese wrote: 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 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] Strange Fromuser behavior?
Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Any idea how this is possible? Kind reagards, E. Versaevel ___ 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] hangup()???
I as a similar problem with this: ignorepat = 9 exten = 9,1,Dial,Zap/g2 exten = 9,2,Congestion What if I pressed 9, called a number, and hanged up before someone replies.. It happened with me more than once that the line is left open, waiting for the other side to hangup (what if there is no other side) .. isn't there a timeout for Congestion ? On Monday 22 November 2004 10:47, Altus Snyman wrote: Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help ___ 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] E100P - Generic (Clone) - :)
I guess it's a hoax. I did get a reply. They answered that they did not have a website or any pictures of the boards they are producing. The website was scheduled to be up last Wednesday but I don't get any replies on e-mails either. Remco On Mon, 22 Nov 2004, Marcelo Pacheco wrote: Em Ter 09 Nov 2004 19:20, Richard Moore escreveu: Hello list , I´m happy to announce our first stable production of a E100P generic boards. The prices are : 50 units = US$ 112,50/unit 50 units = US$ 98,10/unit optional resources: consulting : 32 US$ / server - You need provide root access if needed. ( Only Linux supported ) Freight : FOB TAIWAN We accept VISA and Mastercard. Feel free to contact us. Kind Regards, Richard Moore Asterisk Senior Consultant / Engineer Taiwan Was this a hoax ? I e-mailed him twice over 2 weeks, no answer... Anybody able to contact this guy ? Marcelo Pacheco ___ 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] 4 port ISDN BRI pci card
I am after something similar. I want to be able to use 2 bonded ISDN BRI's and I am not sure what hardware will run with asterisk? Anyone got any ideas? Cheers David Miroslav Nachev wrote: Dear Bartosz, Try this: http://www.junghanns.net/asterisk/page17.html quadBRI PCI ISDN EUR 600,- Best Regards, Miroslav Nachev BJ Hello, BJ I am looking for 4 port ISDN BRI card. BJ I have checked wiki and found one, but they do not show prices BJ for that card. Can somebody advise me which ISDN 4 port card works good BJ with Asterisk, BJ Thank you in advance. BJ Bartosz BJ ___ BJ Asterisk-Users mailing list BJ [EMAIL PROTECTED] BJ http://lists.digium.com/mailman/listinfo/asterisk-users BJ To UNSUBSCRIBE or update options visit: BJhttp://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
RE: [Asterisk-Users] hangup()???
Hi, this call is from? Zap channel, Capi channel or other channel? It is possible that you don't detect well hangup from incoming channel. Regards. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Altus Snyman Enviado el: lunes, 22 de noviembre de 2004 9:48 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] hangup()??? Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.797 / Virus Database: 541 - Release Date: 15/11/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.797 / Virus Database: 541 - Release Date: 15/11/2004 ___ 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] Strange Fromuser behavior?
Hmm, a bit closer, Asterisk seems to do the asterisk@ part only with non numeric usernames, ie [EMAIL PROTECTED] stays [EMAIL PROTECTED] but [EMAIL PROTECTED] turns into [EMAIL PROTECTED] -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens E. Versaevel Verzonden: maandag 22 november 2004 13:27 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' Onderwerp: [Asterisk-Users] Strange Fromuser behavior? Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Any idea how this is possible? Kind reagards, E. Versaevel ___ 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] H323 Problems
Yes I do have the latest CVS version and the 0.7.0 version of openH323. What versions should I be using? Regards Peter Landy -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: 22 November 2004 09:01 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 Problems Peter, Peter Landy wrote: New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Did you apply the OpenH323 patch BEFORE configuring/compiling the library? Regards Peter Landy 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] Phones
What VOIP Phones is everyone using and why? Is the a common phone that seems the work the best? Just wondering. The quality and features implemented in each of the voip phones varies rather dramatically from one manufacturer to another. What works fine in one account (with their expectations) may not be considered acceptable at another account, etc. In _very_ general terms the more expensive the phone, the more time the manufacturer spends doing real life research and regression testing of their pre-release firmware. Likewise, the cheaper the phone the higher the chances that you will be the one doing testing on behalf of the manufacturer. Averages: Grandstream and Snom release firmware versions almost weekly, most of which have one problem or another. Cisco Polycom are closer to quarterly, and although you may find a specific problem or two that might impact an account, the production releases of their firmware tend to be better tested with fewer issues and higher stability. Also, which phones tend to be acceptable to a small group of technical users is usually very different ftom a large group of non-technical business users. Technical folks frequently know where the holes are in their specific implementations and quickly adapt to stepping around those holes in day to day use. Non-technical business users will complain when the transfer key (as an example) does not function the way they think it should. If the phone users are expected to contend with home firewalls/nat boxes (as an example), certain phones will work very well while others fail misserably. Some phones hands-free speaker-phone function very well while others are barely usable. Some offer large displays with directory lookup functions while others don't. Some phones have been designed for large scale deployments with centralized management (eg, firmware upgrades, diagnosing problems) while others require a physical phone visit to accomplish the same. Some have alpha display for callerid while others only have numerical displays. One can actually see/read some displays while setting at your desk while others almost require the user to stand up to see it. Some phone sets are so light-weight they drag across the desk when the handset cord is stretched a little, while others feel and work like analog phones. There are some reviews at www.voip-info.org, however keep in mind that a lot of the phone data is dated and the manufacturer has probably fixed at least some of the negatives shown for specific phones. ___ 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] 4 port ISDN BRI pci card
David Uzzell I want to be able to use 2 bonded ISDN BRI's and I am not sure what hardware will run with asterisk? Anyone got any ideas? I have a couple of customers with two HFC cards working on system access (PTP) mode with no problems whatsoever. The cards have the major advantage of being very cheap! 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] hangup()???
Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help Are you sure that asterisk has _actually_ answered the channel? Or, has it just sensed ringing from the pstn line and is attempting to ring the associated dialplan phone? In other words, you can't hang up on an incoming call that hasn't yet been answered. ___ 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] TDM400 FXO stops handling outgoing calls, but still accepts incoming?
Having read through the Caller ID code, it appears that this is indeed what is
happening. The Caller ID code doesn't contain any logic to trigger a timeout
if no Caller ID data stream is found, or if a stream starts and does not
terminate.
The attached patch causes the Caller ID to timeout after processing around 15
seconds of data. I assume that this should be quite long enough, but I am no
Caller ID expert!
I've tested this patch with BT's automated line test (dial 17070, options 3,
1, 2). It appears to work fine:
== Starting post polarity CID detection on channel 3
-- Starting simple switch on 'Zap/3-1'
Nov 22 13:16:03 NOTICE[7862]: chan_zap.c:5257 ss_thread: Got event 17 (Polarity
Reversal)...
Nov 22 13:16:13 ERROR[7862]: callerid.c:257 callerid_feed: Caller ID processed
120160 samples, giving up.
Nov 22 13:16:13 WARNING[7862]: chan_zap.c:5272 ss_thread: CallerID feed failed:
Success
Nov 22 13:16:13 WARNING[7862]: chan_zap.c:5284 ss_thread: CallerID returned
with error on channel 'Zap/3-1'
Nov 22 13:16:15 WARNING[7862]: chan_zap.c:5293 ss_thread: CID timed out waiting
for ring. Exiting simple switch
-- Hungup 'Zap/3-1'
Mark, how do I go about getting this included in CVS?
Thanks,
Will
On Sun, Nov 21, 2004 at 10:34:27PM +, William R Sowerbutts wrote:
H.
Is it possible that the line is detecting a polarity event, decided that the
line is ringing and started listening for a non-existant V23 data stream, and
then the line has not in fact rung?
This would mark the line as busy (and unable to handle an outgoing call) but
when a call did in fact come in the line would then ring correctly.
I believe BT's automated testing equipment can produce these line conditions.
Will
_
William R Sowerbutts [EMAIL PROTECTED]
Carpe post meridiem http://sowerbutts.com
main(){char*s=#=0 [EMAIL PROTECTED]@^7=,c=0,m;for(;c15;c++)for
(m=-1;m7;putchar(m++/6c%3/2?10:s[c]-311m?42:32));}
--- asterisk/callerid.c.orig2004-11-22 13:01:33.0 +
+++ asterisk/callerid.c 2004-11-22 13:01:59.0 +
@@ -43,6 +43,7 @@
int flags;
int sawflag;
int len;
+ int eaten;
};
@@ -132,6 +133,7 @@
cid-fskd.cont = 0; /* Digital PLL reset */
cid-fskd.x0 = 0.0;
cid-fskd.state = 0;
+ cid-eaten = 0;
memset(cid-name, 0, sizeof(cid-name));
memset(cid-number, 0, sizeof(cid-number));
cid-flags = CID_UNKNOWN_NAME | CID_UNKNOWN_NUMBER;
@@ -249,6 +251,12 @@
ast_log(LOG_WARNING, Out of memory\n);
return -1;
}
+ cid-eaten += len;
+ if(cid-eaten (8000 * 15)){
+ /* we've eaten over 15 seconds of data */
+ ast_log(LOG_ERROR, Caller ID processed %d samples, giving up.\n,
cid-eaten);
+ return -1;
+ }
memset(buf, 0, 2 * len + cid-oldlen);
memcpy(buf, cid-oldstuff, cid-oldlen);
mylen += cid-oldlen/2;
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Re: [Asterisk-Users] A new alternative to see who is online
Hi, googd work, could it be any chance to make it running on linux box ?? JFA On Mon, 2004-11-22 at 09:16 +0100, Nicolas wrote: Thanks for all your comment. I will release the source code but not for the moment. I really need to clean the graphical part but right away I don't have enough time. Regards, Nicolas Bruxer Lyle Giese wrote: 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 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 signature.asc Description: This is a digitally signed message part ___ 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] (Analog Intercom) PagePal by ATT -- was hooked to a Merlin
On Thu, 18 Nov 2004 18:52:21 -0600, Jeb Campbell [EMAIL PROTECTED] wrote: I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of an analog phone to page system, but of course I would like to reuse what is there. Thanks for any advice or pointers, Try this http://www.pagepac.com/pdfs/pagepal.pdf You should be able to connect the system to an analoge port Jason ___ 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] 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 ? -- Clive On Fri, 19 Nov 2004 09:44:13 -0800, Michael Swan [EMAIL PROTECTED] wrote: At 02:55 AM 11/19/2004 +, you wrote: 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 Neon Software, Inc. Hello, I had the same problem with the SipTone - it's just a matter of setting the dtmfmode in the sip.conf file. I think I set it to inband - I remember setting it to either that or rfc2833 or whatever that rfc number is - the correct number is available in the sip.conf fdile itslf. Just fiddle with the dtmf mode - either inband or rfc and u'll be fine. Hope this helps. Shireen Thanks guys. Tried all suggestions above and some of my own. Nothing worked, Tried every combination OF INFO, RFC2833, Inband on phone and in Sip. No good In desparation I reset EVERYTHING to defaul, rebooted, then put all my data back in. IT WORKS ! Must have made a typo or something in the phone setup, but I'm damned if I could find it. Only thing that stopped working now is the VM button, even tho that is set up as per Michaels instructions. I can get at the voicemail by dialling 88 anyway, so I am leaving it alone :-) Thanks again. -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ 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] wiki down ?
im getting: Fatal error: Unknown function: mssql_get_last_message() in /var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on line 415 to the wiki.. Jason ___ 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?
If your device supports Dial Off-Hook or PLAR Code, then you could send a series of unique digits down the line to trigger the AGI script. On Saturday 20 November 2004 12:52 pm, Michael Vogel wrote: 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 -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ 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] E100 or TE410 card an PRA line
Hi, I don't kinow if I'm out of time but.. I in Swizerland (Geneva) and use a E1 which is configured like this: 1) 120 ohmes on RJ45 (which is a standard) 2) digium cards support both but I have CRC4 on 3) sure it's handled two way full IF I can help you in anyway don't hesitate to contact me off line JFA On Fri, 2004-11-19 at 09:14 +0100, Michael Devenijn wrote: We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ 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 signature.asc Description: This is a digitally signed message part ___ 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] E100P - Generic (Clone) - :)
Hello, An hoax is a code that pretend to be malicius but in fact isn't (usualy a virus that do nothiing bad just saying that is there) ragards JFA On Mon, 2004-11-22 at 09:33 -0200, Marcelo Pacheco wrote: Em Ter 09 Nov 2004 19:20, Richard Moore escreveu: Hello list , Im happy to announce our first stable production of a E100P generic boards. The prices are : 50 units = US$ 112,50/unit 50 units = US$ 98,10/unit optional resources: consulting : 32 US$ / server - You need provide root access if needed. ( Only Linux supported ) Freight : FOB TAIWAN We accept VISA and Mastercard. Feel free to contact us. Kind Regards, Richard Moore Asterisk Senior Consultant / Engineer Taiwan Was this a hoax ? I e-mailed him twice over 2 weeks, no answer... Anybody able to contact this guy ? Marcelo Pacheco ___ 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 signature.asc Description: This is a digitally signed message part ___ 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?
Brian Wilkins schrieb: If your device supports Dial Off-Hook or PLAR Code, then you could send a series of unique digits down the line to trigger the AGI script. Now everything works. So I don't need any tricks. Yesterday I wrote a simple dialplan that suits my needs. Yesterday I also wrote a dynamic dial-timeout (per digit) calculation (for typing), depending on the user's input speed. Now I only need an analog modem so I could do a dial-out (and dial-in) over this line as well. 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
Re: [Asterisk-Users] wiki down ?
Hi, Il giorno lun, 22-11-2004 alle 08:49 -0500, Jason p ha scritto: Fatal error: Unknown function: mssql_get_last_message() in /var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on also here...perhaps they're switching away from mssql ? :) Matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [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
RE: [Asterisk-Users] E100P - Generic (Clone) - :) OT
What you describe is a computer hoax. A look in a dictionary will give you this: Main Entry: hoax Pronunciation: 'hOks Function: transitive verb Etymology: probably contraction of hocus Date: circa 1796 : to trick into believing or accepting as genuine something false and often preposterous synonym see DUPE - hoaxer noun or this: Main Entry: hoax Function: noun Date: 1808 1 : an act intended to trick or dupe : IMPOSTURE 2 : something accepted or established by fraud or fabrication Not everything in life is about or related to IT! Yiannis. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of genuix Sent: 22 November 2004 14:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E100P - Generic (Clone) - :) Hello, An hoax is a code that pretend to be malicius but in fact isn't (usualy a virus that do nothiing bad just saying that is there) ragards JFA On Mon, 2004-11-22 at 09:33 -0200, Marcelo Pacheco wrote: Em Ter 09 Nov 2004 19:20, Richard Moore escreveu: Hello list , Im happy to announce our first stable production of a E100P generic boards. The prices are : 50 units = US$ 112,50/unit 50 units = US$ 98,10/unit optional resources: consulting : 32 US$ / server - You need provide root access if needed. ( Only Linux supported ) Freight : FOB TAIWAN We accept VISA and Mastercard. Feel free to contact us. Kind Regards, Richard Moore Asterisk Senior Consultant / Engineer Taiwan Was this a hoax ? I e-mailed him twice over 2 weeks, no answer... Anybody able to contact this guy ? Marcelo Pacheco ___ 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] Grandstream Ringtone
Thanks for the link ___ 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] SER is a better NAT solution?
*bump* (B (Bwoops..this isn't a forum... (B (Bbut anyway, this is a good question. We will soon have somewhere in the 500 (BSIP users range and if I can have 1 machine (SER) handle all the (Bregistration (hopefully out of a database) that will defiantly reduce the (Bload on my asterisk servers. (B (BI've also heard that SER can do load balancing between multiple * servers. (BAny comments/HOWTOs on that? (B (BThanks, (BMatthew (B- Original Message - (BFrom: "Kuniyoshi Murata" [EMAIL PROTECTED] (BTo: [EMAIL PROTECTED] (BSent: Sunday, November 21, 2004 6:24 PM (BSubject: [Asterisk-Users] SER is a better NAT solution? (B (B (B Hi, (B (B I'm now setting up a VoIP conference room using Asterisk. (B (B All the clients are SIP phone (to be exact, Xlite), number of clients that (Bshould be registered are around 50 and concurrent users are maybe 15 clients (Bat most. (B (B So, basically I think I can handle the situation only with Asterisk. (B I'm wondering however, most of my clients are behind NAT of home router (Band using SER together with Asterisk sounds better solution. (B (B Is there significant difference between SER and Asterisk, for handling SIP (Bphones behind NAT? (B (B -- (B Kuniyoshi Murata.iChat/AIM:macwebcaster (B English-Japanese Interpreter mailto:[EMAIL PROTECTED] (B Macintosh Webcast Specialisthttp://www.macwebcaster.com (B ___ (B Asterisk-Users mailing list (B [EMAIL PROTECTED] (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list ([EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test Number in the UK?
In recent times I have seen a few posts which describe how to use ztmonitor to set the rxgain and txgain parameters on an FXO channel. The starting point is to call a '102 milliwatt test number' Does anybody know of such a facility in the UK? Ian ___ 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] incompatible with our capability 0x400.
On Sun, 21 Nov 2004 18:54:25 +0500, khurram bhatti [EMAIL PROTECTED] wrote: I'm trying to connect * server from diax 0.9.8c client and * outputs this errors on CLI Nov 21 18:59:59 NOTICE[7316]: chan_iax2.c:5742 socket_read: Rejected connect attempt from 192.168.0.4, requested/capability 0x2/0x2 incompatible with our capability 0x400. You have a codec problem your * only supports 0x400 (ILBC) the diax requested 0x2 (GSM) so no compatible codec is availiable Jason ___ 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] callprogress option
From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? Thanks, Shaun Tierney ___ 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] SER is a better NAT solution? Addendum: Linksys WRT54G
I am quite interested in this as well. I didn't realize registrations are the #1 cause of load on an asterisk server, we haven't gotten to that kind of usage just yet. People were having problems with compiling Asterisk on a hacked Linksys WRT54G, issues with compiling against uClibc and some threading issue. However, has anybody tried to compile SER in/on one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, November 22, 2004 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SER is a better NAT solution? *bump* woops..this isn't a forum... but anyway, this is a good question. We will soon have somewhere in the 500 SIP users range and if I can have 1 machine (SER) handle all the registration (hopefully out of a database) that will defiantly reduce the load on my asterisk servers. I've also heard that SER can do load balancing between multiple * servers. Any comments/HOWTOs on that? Thanks, Matthew - Original Message - From: Kuniyoshi Murata [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 21, 2004 6:24 PM Subject: [Asterisk-Users] SER is a better NAT solution? Hi, I'm now setting up a VoIP conference room using Asterisk. All the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most. So, basically I think I can handle the situation only with Asterisk. I'm wondering however, most of my clients are behind NAT of home router and using SER together with Asterisk sounds better solution. Is there significant difference between SER and Asterisk, for handling SIP phones behind NAT? -- Kuniyoshi Murata.iChat/AIM:macwebcaster English-Japanese Interpreter mailto:[EMAIL PROTECTED] Macintosh Webcast Specialisthttp://www.macwebcaster.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 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] Fw: TDMoE over bonded NIC's
On Sun, 21 Nov 2004 19:50:36 -, Kevin Brennan [EMAIL PROTECTED] wrote: I am planning to configure * box A with PSTN interface to route faxes to * box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for connection between servers. Was wondering - does anybody have experience with TDMoE over bonded interface - ie. does it work ok?. - does anybody have feedback using this scenario for fax? another question, perhaps someone knows what's the limitation for channels on TDMoE interface ? and is there a workaround. I recommend you use Iax trunking rather than TDMoE this would scale better. Jason ___ 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] Call Deflection (CD) with ZapHFC
Hi to everybody, is it possible to use ISDN Call Deflection with a ZapHFC card? Regards Bastian ___ 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] which ISDN Card?
Hello, I am a newbie with asterisk; I´ve searching the mailinglist, www.voip-info.org, isdn4linux web... But I don´t know which isdn card to buy. I want the * box to be able to send faxes, and obviusly to send and receive calls. 1) What do you recomend me? 2) Would AVM ISDN Fritz Card PCI V2.0 work? and Eicon Diva ISDN Modem PCI? 3) Do you know any cheap site to buy? Many 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] A new alternative to see who is online
Hi, You are true for the moment, I only listen to messages and don't request anything to asterisk. So I only manage registration when a sip phone register or unregister while the leon is already launched. Two possible solutions: -Havin a server that remember states which I want to avoid -asking asterisk when leon connects.(I didn't had time to explore this solution) Jens Hansen wrote: Great program, thanks! only one question: when i reboot my pc i can't see who is online, until the sip user re-register their clients at the server. leon only seems to update his online status when a sip client connects. Jens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Sent: Monday, November 22, 2004 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] A new alternative to see who is online Thanks for all your comment. I will release the source code but not for the moment. I really need to clean the graphical part but right away I don't have enough time. Regards, Nicolas Bruxer Lyle Giese wrote: 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 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
Re: [Asterisk-Users] A new alternative to see who is online
Hi, Sorry it is not planned. Regards, Nicolas genuix wrote: Hi, googd work, could it be any chance to make it running on linux box ?? JFA On Mon, 2004-11-22 at 09:16 +0100, Nicolas wrote: Thanks for all your comment. I will release the source code but not for the moment. I really need to clean the graphical part but right away I don't have enough time. Regards, Nicolas Bruxer Lyle Giese wrote: 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 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
Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Fw: TDMoE over bonded NIC's
On Mon, 22 Nov 2004, Jason Williams wrote: I recommend you use Iax trunking rather than TDMoE this would scale better. Using iax trunking will also loose the advantage of being tdm all the way, i.e. low latancies. If the rest of the setup is tdm there is a lot of value in not going to voip for one hop. Peter ___ 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: Snom 190 - dhcp - settings_server
On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote: However I would use a more specific path for a web-server ;-) Something like: option tftp-server-name http://192.168.0.9/snom/snom200.htm But for the snom 190 tftp-server-name in dhcp config will set update_server. The field/variable setting_server remains empty. The documentation suggests that dhcp data can be used to define setting_server. Just a bug in the snom 190 firmware ? Best regards -- Stefan Tichy [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
Re: [Asterisk-Users] which ISDN Card?
I have no experience with faxes * at all but the AVM ISDN Fritz Card PCI V2.0 works very well for me (with SuSE 9.1) and I found them easy to get on ebay.de Derek Rubens Sanchez wrote: Hello, I am a newbie with asterisk; I´ve searching the mailinglist, www.voip-info.org, isdn4linux web... But I don´t know which isdn card to buy. I want the * box to be able to send faxes, and obviusly to send and receive calls. 1) What do you recomend me? 2) Would AVM ISDN Fritz Card PCI V2.0 work? and Eicon Diva ISDN Modem PCI? 3) Do you know any cheap site to buy? Many 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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] Unknown number CID on SIP phone
Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway, withough the dots, so if my gateway is at 10.0.0.1, Asterisk reports a call from 10001 instead of reporting Unknown, or simply not reproting anything at all. It looks like there was some dicussions about a caller ID translation table. Is something like that what would be needed? Thanks, Brian ___ 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] Linksys RT31P2
Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652scid=29 -Ron___ 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] Creating CDR's with online connected time
Hi there, How do i setup asterisk, so that in the CDR's is only the time, which the line actually was connected? Not the time, the line was up, but the time the user was able to talk to another user. Thanks in advance, Carsten ___ 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] Cisco Call Manager and Asterisk
Has anyone been successful interfacing call manager and Asterisk? ___ 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] hangup()???
Altus Snyman wrote: Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing This is the way it works with analog ports. Asterisk should realize it's no longer getting ring voltage after about 5 seconds or so. ___ 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] Linksys RT31P2
If anyone finds the generic version of this available (i.e., not locked to Vonage), please advise the list of where. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 22 Nov 2004 [EMAIL PROTECTED] wrote: Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652scid=29 -Ron This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-22%5C543f125be9b24494a8d7fa465e02817cC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] SPA-841 / SPA-2100 Canadian Distributor
Try George at www.netvoice.ca - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 21, 2004 9:26 PM Subject: [Asterisk-Users] SPA-841 / SPA-2100 Canadian Distributor Does anybody know Canadian Distributor for SPA-841 and SPA-2100 -- #Joseph ___ 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] IAX error tolerence??
WipeOut wrote: Hi, Didn't get any opinions on the log file I mailed onto the list over the weekend so I am continuing to try and track the cause for the dropped calls.. I have a feeling that its to do with IAX being way too sensitive when it comes to packet loss.. Since it is going across the internet it needs to be more tolerant when it comes to errors and packet loss.. Are there any settings (in the conf file or the source) that I can change to make it more able to keep the call connected without dropping it when there are errors even if the sound breaks up for a couple of seconds?? There's nothing in chan_iax2 that should cause a call to be dropped if there's packet loss for just a couple of seconds. I think a call is not dropped until retransmission of reliable frames fails 4 times, if this happens, you should see something like this in your log: Max retries exceeded to host %s on %s (type = %d, subclass = %d, ts=%d, seqno=%d)\n, Each retry is sent 10 times later than the first retry (up to 10 seconds). The first retry is sent at twice the round-trip time between the two endpoints. -SteveK ___ 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] H323 Problems
Peter Landy wrote: Yes I did. Does anyone have a working list of libraries and versions. I have tried with different releases of H323 and they all give different errors. Also is it necessary to compile the H323 under asterisk src/channels/H323 as this also bails on errors. The rest of my asterisk install works perfectly it is just H323 I am hung up on. There is a README in asterisk src/channels/H323 that tells you which versions you need. ___ 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] which ISDN Card?
[EMAIL PROTECTED] is believed to have said: Hello, I am a newbie with asterisk; I¥ve searching the mailinglist, www.voip-info.org, isdn4linux web... But I don¥t know which isdn card to buy. I want the * box to be able to send faxes, and obviusly to send and receive calls. 1) What do you recomend me? 2) Would AVM ISDN Fritz Card PCI V2.0 work? and Eicon Diva ISDN Modem PCI? 3) Do you know any cheap site to buy? Many thanks. Rubens, where are you located? I have just ordered from an italian reseller the AVM Fritz Card, for about 70 Euros. As I could not locate any reseller info I just wrote to AVM (from their website, http://www.avm.de/en/index.php3) and got a couple of references to local dealers very quickly. AFAIK the Fritz Card is a popular and cheap ISDN solution (at least for Europe). I will see in a couple of days how the jump from theory to reality really looks like. HTH Aldo ___ 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] Linksys RT31P2
On November 22, 2004 10:47 am, [EMAIL PROTECTED] wrote: Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. How well do these Geode and ARM-based systems handle VPN anyway? I would have figured you would want a decent processor to handle more than maybe one or two clients. -A. ___ 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: Re: i swtiched to digest
On November 19, 2004 05:17 pm, FuturaHost.Com Lists wrote: Yes and no would suffice, so we can close this without a talk long a year, and without someones forcing their point of view to others. Sorry but yes or no does not suffice because the very next post will be Why? -A. ___ 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] TDM400 FXO stops handling outgoing calls, but still accepts incoming?
Hi Will, snip Having read through the Caller ID code, it appears that this is indeed what is happening. The Caller ID code doesn't contain any logic to trigger a timeout if no Caller ID data stream is found, or if a stream starts and does not terminate. /snip I submitted a patch for this which was included in the the CVS build of the 19th of November. See bug http://bugs.digium.com/bug_view_page.php?bug_id=0002909 Regards Ian ___ 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] callprogress option
Shaun Tierney wrote: From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? If you ask me it doesnt' work well nomatter what kind of line you have. VoIP (IAX/SIP/H323), PRI, and T-1/E-1 do not need callprogress. The telco provides everything required for progress detection. Analog ports don't (usually) provide this and so callprogress is needed. ___ 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 and H.323 Gatekeeper
Thank you, I will see into it. Regards, Jorge A. -Mensaje original- De: Paul Davidson [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 12:12 PM Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Unpredictables Hangups
-Original Message- From: Stefano Finetti [mailto:[EMAIL PROTECTED] Sent: November 19, 2004 6:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unpredictables Hangups Michael, I just check'd my kernel configuration... I have APIC support and no Enhanced Real Time Clock, exactly as you have on your hardware. It *could* be a timer issue, except that i can't manage how to accelerate mi timer or to slow down my t1xxp driver... I believe the timer frequency is controlled by the value of HZ defined in linux/param.h when compiling the kernel. You'll see it as 100 on some kernels and 1024 on others (particularly Redhat). See: http://www.linuxgazette.com/node/view/8993 Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ 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] callprogress option
Ok, so if I turn callprogress off, and try to connect a call which is bridged between an incoming line and an outgoing line, will it treat the call as being answered once it is bridged or once it is actually answered on the outgoing T1 trunk? Thanks, Shaun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: Monday, November 22, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] callprogress option Shaun Tierney wrote: From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? If you ask me it doesnt' work well nomatter what kind of line you have. VoIP (IAX/SIP/H323), PRI, and T-1/E-1 do not need callprogress. The telco provides everything required for progress detection. Analog ports don't (usually) provide this and so callprogress is needed. ___ 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] Linksys RT31P2
snip How well do these Geode and ARM-based systems handle VPN anyway? I would have figured you would want a decent processor to handle more than maybe one or two clients. /snip These are really designed for home use. I use the BEHVP41 to keep ~ 10 VPN tunnels open. Works great. Not sure at what point the processor may become an issue... ___ 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: Asterisk-Users Digest, Vol 4, Issue 298
Yes, I have both Call Manager and Call Manager Express integrated with *. Prior to Call Manager 4.0 you would need to perform an H.323 integration with *. As of CM 4.0 Cisco supports SIP trunking so this would be the preferred method of integration. This config is on http://www.voip-info.org Seems like the site is having problems now otherwise I would have provided the direct link. I also have Call Manager Express integrated with * using SIP trunking. As soon as I get my configs cleaned up I'll post them on the Wiki. Keith Date: Mon, 22 Nov 2004 10:56:53 -0500 From: Aster risk [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco Call Manager and Asterisk To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Has anyone been successful interfacing call manager and Asterisk? ___ 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] Linksys RT31P2
Andrew Kohlsmith wrote: How well do these Geode and ARM-based systems handle VPN anyway? I would have figured you would want a decent processor to handle more than maybe one or two clients. The CPU is only a limitation for a VPN if the pipe the VPN is running over is large/wide. These devices are typically used at the end of a DSL/cable connection, with a maximum bandwidth of a few megabits per second. I don't think that a 200MHz Geode or ARM will have any trouble keeping up with that amount of traffic. ___ 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] Analog ports via USB
I think the USB IP Phone adaptor is a S100U - I found the TigerJet website/products by reading the chip inside a S100U that I purchased at digium and they look identical - but dont trust me on this - I didn't buy one from TigerJet direct. Derek Michael Vogel wrote: Derek Conniffe schrieb: Re: the S100Us - I think you can get them from www.tjnet.com (TigerJet). You are probably after their USB to RJ11 adapter. I think that the Personal Phone Gateway-PCI cards are generic X100Ps too Do you know if the USB phone and the USB IP Phone adaptor is Linux compatible? 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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] Re: Firefly Problems
Chris Olson wrote: Hello, I have firefly installed and it is somewhat working. It is registering with my Asterisk server and I can call out, but I receive no audio coming into Firefly. From the Asterisk end, everything looks OK with the call, just no audio is being received on the Firefly end. I am using 1.9.6 Any ideas? a fix for this will be out tommorrow - you can temporarily fix it by inserting the r option into your dial cmd cheers, Adam Thanks Adam. Can you let us know when the fix is available and where we can download the fixed 3rd-party from? A little more info ... this is actually a one-way audio problem as audio passes from Firefly to Asterisk, but not from Asterisk to Firefly. ___ 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 update
On Fri, 2004-11-19 at 18:01, Kevin wrote: I have applied the revised patch. After working through the steps to follow (I think there may be another mistake in the steps) I get a busy when calling out: -- Got SIP response 404 Not Found back from 147.135.0.128 -- SIP/sip.broadvoice.com-39f6 is circuit-busy When I change the host=proxy.dca.broadvoice.com (which is my closest) to sip.broadvoice.com it works. Is there a typo in the instructions? I am getting the same error on outbound calls although incoming calls work with the new patch and configuration. Making the change you describe fixed things for me. I can't imaging that it was a typo though because the whole section about choosing a proxy is specifically for setting the host= section in the config file. Any ideas about the above error? I'm very nervous about my current BV setup since it's not the official configuration. I'm just waiting for it to break. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] app_sms: problems sending a sms
Has this worked finally? Can you send me the configs if they indeed
have been working.
Seshu kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steffen
Koepf
Sent: Friday, November 19, 2004 8:40 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] app_sms: problems sending a sms
Hello,
i try to send out a sms, but with no success.
The trunk is a E100P, and the sms should go out to the Telekom SM-SC.
What i want to to at the first run is, sending out a sms when a certain
number is dialed.
I tried:
In extensions.conf:
exten = 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,Hi there) exten
= 35953,2,SMS(${TRUNK}/9350193010) exten = 35953,3,Hangup
exten = 35954,1,Dial(${TRUNK}/9350193010)
and get:
tkserv*CLI
-- Executing Goto(SIP/35903-da57, voiplocal|35953|1) in new
stack
-- Goto (voiplocal,35953,1)
-- Executing SMS(SIP/35903-da57,
Zap/g1/9350193010||0179NUMBER|Hi there) in new stack
-- Executing SMS(SIP/35903-da57, Zap/g1/9350193010) in new stack
-- SMS TX 92 01 FF 6E 00 00...
-- Executing Hangup(SIP/35903-da57, ) in new stack
== Spawn extension (voiplocal, 35953, 3) exited non-zero on
'SIP/35903-da57'
935 is the prefix to go out to the world via a telekom PRI line.
Sometimes i hear a chirp like the sound of a bird, sometimes i get this
SMS TX 92 01 FF 6E 00 00... line, sometimes nothing happens but a
hangup after a few seconds.
(0179NUMBER is the number of the cell-phone).
When i call the 35954 via a SIP Phone, i hear always one chirp, and a
hangup after a few seconds, so i guess the call reaches the SM-SC.
Does someone know whats wrong?
cu,
Steffen
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[Asterisk-Users] Problem with fax tone (CNG) from TxFax and busy detect
I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: filename.tiff|caller Using the |caller parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable to detect the busy tones. If I were to remove |caller then the receiving station wouldn't receive the CNG tone and possibly not direct the call to the fax machine. Is there a way for * to detect busy tones while ignoring (filtering) the fax tones? Thanks, -- 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] Siemens optiPoint 300
Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to them in the archives of this list, and the Wiki seems to be down. I have a chance to pick up a bunch of these, cheap. Questions: * Asterisk support? * What sort of power supplies will they need? The bunch I am looking at are surplus and have no supplies. Thanks, /edg Ed Greenberg San Jose, 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
Re: [Asterisk-Users] Re: Snom 190 - dhcp - settings_server
On Monday 22 November 2004 16:01, Stefan Tichy wrote: On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote: However I would use a more specific path for a web-server ;-) Something like: option tftp-server-name http://192.168.0.9/snom/snom200.htm But for the snom 190 tftp-server-name in dhcp config will set update_server. The field/variable setting_server remains empty. The documentation suggests that dhcp data can be used to define setting_server. Just a bug in the snom 190 firmware ? no, for the dhcp option 66 its ok like that ! Its internally handled correctly. Sven Best regards -- --- See our FAQs at: http://www.snom.com/faq_en.php --- snom technology AG Pascalstraße 10b D-10587 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.comsip:[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] Zap - 256 format frames
Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto(SIP/3044-8d49, cytel-outgoing|915124512424|1) in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum(SIP/3044-8d49, 2814494000) in new stack -- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new stack -- Called g1/15124512424 -- Zap/1-1 answered SIP/3044-8d49 -- Executing Dial(Zap/2-1, SIP/[EMAIL PROTECTED]|60) in new stack -- Called [EMAIL PROTECTED] -- Accepting call from '2814494000' to '5124512424' on channel 0/2, span1 -- Got SIP response 302 Moved Temporarily back from XXX.XXX.XXX.70 -- Now forwarding Zap/2-1 to 'SIP/[EMAIL PROTECTED]:5060'(thanks to SIP/RNK-1050) -- SIP/XXX.XXX.XXX.52:5060-d8b1 is making progress passing it to Zap/2-1 Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot handle frames in 256 format Nov 22 10:59:32 WARNING[1126867776]: app_dial.c:358 wait_for_answer: Unable to forward frame == Spawn extension (all-incomming, 5124512424, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' sip.conf --- [RNK] snip disallow=all allow=g729 extensions.conf exten = 5124512424,1,Dial(SIP/[EMAIL PROTECTED],60) Thanks, Matthew ___ 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] Problem with fax tone (CNG) from TxFax and busydetect
I'm losing call files in /var/spool/asterisk/outgoing because * isn't able to detect the busy signal. The call file looks like this: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: filename.tiff|caller Using the |caller parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable to detect the busy tones. If I were to remove |caller then the receiving station wouldn't receive the CNG tone and possibly not direct the call to the fax machine. Is there a way for * to detect busy tones while ignoring (filtering) the fax tones? I have a similar problem: I dial out but no CNG signal is sent at all. It is as if the auto-dial program never sees the line has been answered. It never calls the application. I set it up to dial my cell phone. I answer and nothing, dead. I hang up and the script hangs up. My call file is set up as yours. John ___ 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] Problem with fax tone (CNG) from TxFax and busy detect
Michael Welter wrote: Using the |caller parameter, TxFax injects the fax tone (CNG) onto the line. With the CNG tone, asterisk is unable to detect the busy tones. If I were to remove |caller then the receiving station wouldn't receive the CNG tone and possibly not direct the call to the fax machine. Is there a way for * to detect busy tones while ignoring (filtering) the fax tones? There might be a way to patch asterisk to do this, but currently the only way that I know of is to use a PRI or other type of line that provides busy/answer/etc indications. ___ 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] Zap - 256 format frames
Matthew Boehm wrote: -- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new stack Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot handle frames in 256 format show codecs in the Asterisk CLI will tell you the number of each codec. If you want to use G729 and the t or T option on the Dial() line you must purchase the Digium G729 codec. Asterisk CANNOT do pass-thru when using t or T options on the Dial() line. There is no way around this. ___ 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] IPv6 and Asterisk?
Hi, I've been experimenting with an IPv4 and IPv6 VoIP setup using SER. I'm using Asterisk for voicemail, etc. but as this only works for IPv4, I had to do a number of tricks to get it going for IPv6 phones. I was wondering whether there is any interest or plans in the pipeline to have Asterisk IPv6-enabled. Any info, especially by the developers out there, would be welcome. Thanks, -- Socrates Varakliotis UCL Computer Science +44 20 7679 3696 ___ 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] Zap - 256 format frames
I guess I should have mentioned that I have 10 codecs: 0/0 encoders/decoders of 10 licensed channels are currently in use Any other ideas? -Matthew - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:32 AM Subject: Re: [Asterisk-Users] Zap - 256 format frames Matthew Boehm wrote: -- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new stack Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot handle frames in 256 format show codecs in the Asterisk CLI will tell you the number of each codec. If you want to use G729 and the t or T option on the Dial() line you must purchase the Digium G729 codec. Asterisk CANNOT do pass-thru when using t or T options on the Dial() line. There is no way around this. ___ 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] Unknown number CID on SIP phone
Brian McCrary wrote: Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway, withough the dots, so if my gateway is at 10.0.0.1, Asterisk reports a call from 10001 instead of reporting Unknown, or simply not reproting anything at all. You should be able to set the inbound callerid from the switch/gateway to a specific unknown in sip.conf file with just a callerid= line. The place I looked on the wiki didn't show a specific description for the callerid= line, but that's what I thought I read for it somewhere. http://www.voip-info.org/wiki-Asterisk+config+sip.conf (currently hosed) http://64.233.179.104/search?q=cache:IIOmLeG89KwJ:www.voip-info.org/wiki-Asterisk%2Bconfig%2Bsip.conf+site:voip-info.org+sip.conf (google cache) -- Andrew Thompson http://aktzero.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] asterisk manager api to stop a stream file command in an agi
All, I was wondering if it is possible to use the manager api to stop a "stream file" agi command for a channel. Either through posting a DTMF digit to the channel or something like that - or a cleaner way also. My AGI cannot cancel the playing of a "Stream file" command unless the user hits a DTMF key, however, I want to be able to change from one stream file to another stream file. 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
Re: [Asterisk-Users] TDM400 FXO stops handling outgoing calls, but still accepts incoming?
On Mon, Nov 22, 2004 at 04:09:21PM -, Ian D. Wlloughby wrote:
I submitted a patch for this which was included in the the CVS build of
the 19th of November.
See bug http://bugs.digium.com/bug_view_page.php?bug_id=0002909
Ian,
The code I am running includes this patch already (I checked it out from CVS
on the 21st), and it does not seem to resolve the problem on my line.
The patch I have posted earlier today, which causes the Caller-ID code to
abort after 15 seconds, does resolve the problem.
I think applying both patches should be fine -- the belt and braces approach!
I'll append my patch to bug 0002909.
Thanks,
Will
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[Asterisk-Users] edirecting calls with Asterisk
Hello, I am trying a couple of days before to set up asterisk to redirects an incoming call if the extension dialed is busy without success. I just try to use 'Gotoif' command, with bad luck, it can't do what i want. Anybody could helpme? ani clue will be appreciated. Regards. Ismael. ___ 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] TDM400 FXO stops handling outgoing calls, but still accepts incoming?
Ian, The code I am running includes this patch already (I checked it out from CVS on the 21st), and it does not seem to resolve the problem on my line. The patch I have posted earlier today, which causes the Caller-ID code to abort after 15 seconds, does resolve the problem. Hmmm, it should bail out when you get Event 17. This was the patch I put in if you get an event then bail. Here is my log:- Nov 21 16:19:04 VERBOSE[1974]: == Starting post polarity CID detection on channel 4 Nov 21 16:19:04 NOTICE[28670]: Got event 17 (Polarity Reversal)... Nov 21 16:19:06 WARNING[28670]: CID timed out waiting for ring. Exiting simple switch Which looks the same as yours so I am confuse as to why yours isn't bailing as the code is in the same if block as the Got event 17. Are you using V23 and Polarity Reversal as this is the only bit of code I changed, the Bell stuff already had conditions to allow it to bail out. R's Ian ___ 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] edirecting calls with Asterisk
Read up on SetGroup and CheckGroup. On Nov 22, 2004, at 9:57 AM, ismaelg wrote: Hello, I am trying a couple of days before to set up asterisk to redirects an incoming call if the extension dialed is busy without success. I just try to use 'Gotoif' command, with bad luck, it can't do what i want. Anybody could helpme? ani clue will be appreciated. Regards. Ismael. ___ 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] edirecting calls with Asterisk
On Mon, 2004-11-22 at 18:57 +0100, ismaelg wrote:
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i want.
Anybody could helpme?
ani clue will be appreciated.
Did you bother reading the sample configs? I'll admit the copy I am
pulling from is a little old, but in the example config I see this
example. If you can't figure out how to do what you need from this...
you will have a very rough time with anything else.
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] Using IPKall and SIP with insecure=very
Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip debugging turned on I see the call come in and the message below is printed. If I put the exten route that I have in the ipkall-inbound section of extensions.conf (below) into the default section it works fine, but isn't neat and elegant. How do I make incoming call from ipkall match a sip.conf section? Thanks, Rob On the CLI with sip debugging turned on: Found no matching peer or user for '66.54.140.46:5060' Looking for 3501 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found From sip.conf: [3501] type=peer host=dynamic dtmfmode=rfc2833 context=ipkall-inbound insecure=very nat=no From extensions.conf: [ipkall-inbound] exten = 3501,1,Goto(menu,s,1) ___ 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] Cisco 7960 version 7.3 SIP not always able to hear calling person
I have the Cisco 7960 SIP version 7.3 phone. When someone calls in I cannot always hear that person. They can hear me though. (The ear piece is DEAD quite like it is muted or something - no noise at all). This never happens with the other 4 grandstream SIP phones I have. Is there a problem in my setup? Is there a problem with this version of cisco SIP? Any ideas? or is this happening to other users of this phone also? 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
Re: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able to hearcalling person
Using 7.3 here on a 7960 and no problems. Matthew - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 22, 2004 12:51 PM Subject: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able to hearcalling person I have the Cisco 7960 SIP version 7.3 phone. When someone calls in I cannot always hear that person. They can hear me though. (The ear piece is DEAD quite like it is muted or something - no noise at all). This never happens with the other 4 grandstream SIP phones I have. Is there a problem in my setup? Is there a problem with this version of cisco SIP? Any ideas? or is this happening to other users of this phone also? 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 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] Using IPKall and SIP with insecure=very
Rob Emanuele [EMAIL PROTECTED] wrote: I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip debugging turned on I see the call come in and the message below is printed. If I put the exten route that I have in the ipkall-inbound section of extensions.conf (below) into the default section it works fine, but isn't neat and elegant. How do I make incoming call from ipkall match a sip.conf section? From sip.conf: [3501] type=peer host=dynamic dtmfmode=rfc2833 context=ipkall-inbound insecure=very nat=no From extensions.conf: [ipkall-inbound] exten = 3501,1,Goto(menu,s,1) You'll probably find that there's no need to set up a specific user for IPKall. You were using type = peer, which would have been wrong anyway. In your [general] section, create a context = incoming-sip (or whatever you want to call it) and then set up a matching context in extensions.conf. Your extensions.conf context can then match your 3501 extension, along with any other direct incoming SIP addresses you need. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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
Re: [Asterisk-Users] IAXy Configuration
On Fri, 2004-11-19 at 15:31 -0800, Erik Espinoza wrote: 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 I remember a note on the list about issues with a cisco switch, and conecting an iaxy. Mine wouldn't grab an ip either (win2k server), and a cisco 3500. I haven't had time to try a different switch yet. -- Tony Nichols [EMAIL PROTECTED] Appalachian Log Structures Inc. ___ 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] Using IPKall and SIP with insecure=very
Rob Emanuele [EMAIL PROTECTED] wrote: I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip debugging turned on I see the call come in and the message below is printed. If I put the exten route that I have in the ipkall-inbound section of extensions.conf (below) into the default section it works fine, but isn't neat and elegant. How do I make incoming call from ipkall match a sip.conf section? From sip.conf: [3501] type=peer host=dynamic dtmfmode=rfc2833 context=ipkall-inbound insecure=very nat=no From extensions.conf: [ipkall-inbound] exten = 3501,1,Goto(menu,s,1) You'll probably find that there's no need to set up a specific user for IPKall. You were using type = peer, which would have been wrong anyway. In your [general] section, create a context = incoming-sip (or whatever you want to call it) and then set up a matching context in extensions.conf. Your extensions.conf context can then match your 3501 extension, along with any other direct incoming SIP addresses you need. What if I wanted to create different incoming-sip contexts depending on the service being used or number being called? For example sip calls coming from ipkall goto one context that presents a menu, but another sip call coming from one of the free German services provides a different menu and in German. Thanks, Rob ___ 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] Zap - 256 format frames
Well, it seems that Zap cannot do 729 at all:
channels/chan_zap.c (line 4156):
if ((frame-subclass != AST_FORMAT_SLINEAR)
(frame-subclass != AST_FORMAT_ULAW)
(frame-subclass != AST_FORMAT_ALAW)) {
ast_log(LOG_WARNING, Cannot handle frames in %d format\n,
frame-subclass);
return -1;
}
Seems that Zap can only do slinear, ulaw and alaw. So, how do I force my zap
card to only to alaw?
If I have an incomming Sip 729 call, shouldn't asterisk convert it to 711 to
go out the zap card? I have 10 g729 licenses.
-Matthew
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 22, 2004 11:24 AM
Subject: [Asterisk-Users] Zap - 256 format frames
Any ideas on this warning? If I call this number, sometimes I get this
error
and sometimes the call goes thru fine. Why would it work sometimes?
-- Executing Goto(SIP/3044-8d49, cytel-outgoing|915124512424|1) in
new stack
-- Goto (cytel-outgoing,915124512424,1)
-- Executing SetCIDNum(SIP/3044-8d49, 2814494000) in new stack
-- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new
stack
-- Called g1/15124512424
-- Zap/1-1 answered SIP/3044-8d49
-- Executing Dial(Zap/2-1, SIP/[EMAIL PROTECTED]|60) in new stack
-- Called [EMAIL PROTECTED]
-- Accepting call from '2814494000' to '5124512424' on channel 0/2,
span1
-- Got SIP response 302 Moved Temporarily back from XXX.XXX.XXX.70
-- Now forwarding Zap/2-1 to
'SIP/[EMAIL PROTECTED]:5060'(thanks to SIP/RNK-1050)
-- SIP/XXX.XXX.XXX.52:5060-d8b1 is making progress passing it to
Zap/2-1
Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot
handle
frames in 256 format
Nov 22 10:59:32 WARNING[1126867776]: app_dial.c:358 wait_for_answer:
Unable
to forward frame
== Spawn extension (all-incomming, 5124512424, 1) exited non-zero on
'Zap/2-1'
-- Hungup 'Zap/2-1'
sip.conf
---
[RNK]
snip
disallow=all
allow=g729
extensions.conf
exten = 5124512424,1,Dial(SIP/[EMAIL PROTECTED],60)
Thanks,
Matthew
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RE: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able tohearcalling person
Hello, I have a similar issue with the PingTel xpressa: audio is not sent from the phone to *. Has anyone else experienced it? Best, Alessandro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, November 22, 2004 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able tohearcalling person Using 7.3 here on a 7960 and no problems. Matthew - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 22, 2004 12:51 PM Subject: [Asterisk-Users] Cisco 7960 version 7.3 SIP not always able to hearcalling person I have the Cisco 7960 SIP version 7.3 phone. When someone calls in I cannot always hear that person. They can hear me though. (The ear piece is DEAD quite like it is muted or something - no noise at all). This never happens with the other 4 grandstream SIP phones I have. Is there a problem in my setup? Is there a problem with this version of cisco SIP? Any ideas? or is this happening to other users of this phone also? 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 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] Zap - 256 format frames
Matthew Boehm wrote:
Well, it seems that Zap cannot do 729 at all:
channels/chan_zap.c (line 4156):
if ((frame-subclass != AST_FORMAT_SLINEAR)
(frame-subclass != AST_FORMAT_ULAW)
(frame-subclass != AST_FORMAT_ALAW)) {
ast_log(LOG_WARNING, Cannot handle frames in %d format\n,
frame-subclass);
return -1;
}
Seems that Zap can only do slinear, ulaw and alaw. So, how do I force my zap
card to only to alaw?
If I have an incomming Sip 729 call, shouldn't asterisk convert it to 711 to
go out the zap card? I have 10 g729 licenses.
Correct. Zap can only do slinear, ulaw, and alaw. Asterisk will
transcode from whatever codec you are using for the VoIP leg of the
call into ulaw or alaw. Since you have G729 licenses this should
happen automagically. I have no idea why this is happening.
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