Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
We're currently running a PSTN - SIP gateway with Asterisk. We also run IAX/SIP - PSTN. We have performed a test where the call is routed UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... I did ask them to turn down the speakers and retest and it appears the echo was still there. I don't quite understand how the echo is being introduced in this case. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Authentication bug?
Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan
RE: [Asterisk-Users] Music on hold Read error on sound device
My musiconhold.conf is as below: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 random = quietmp3:/var/lib/asterisk/mohmp3,-z This has been copied from the working system as has the mp3 file into /var/lib/asterisk/mohmp3. The mp3 file also has the same permissions as the working system. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: 21 July 2003 01:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device Stuart Hirst wrote: When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? What does your musiconhold.conf file look like? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on hold Read error on sound device
This is a recent CVS checkout and show version reports Asterisk CVS-07/19/03-22:42:04 What's alsa. I have not come across that yet. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: 21 July 2003 02:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device You didn't mention the distro you are using. I'm wondering if you are using one of the distros that leans towards the alsa drivers. If so, then chan_oss would have problems. On Sun, 2003-07-20 at 19:16, Stuart Hirst wrote: I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable I have checked that the sound card works by loading X11 and running sound tests which is fine. I have used lsof /dev/dsp to see if another application or server is controlling the sound device and without * running, nothing is reported. With * running lsof reports that * has the device. Voicemail works fine. When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? Rgds, Stuart -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones
Hello all, I am a newbie to this list - and so far very impressed with the functionality of Asterisk. So far I have setup a simple soft phone running on a windows PC making calls to other SIP soft phones. Later this week I hope to get UK ISDN2e up and running with it! My question is I would like the experience and feedback from users about what equipment/software you are all using for phones to connect to Asterisk, so fat I have been playing with xten soft phone which works very well but before I make my decision on which phones to use would like the feedback of the group. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best E1 channel bank?
Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run your E-1 connection to the telco and run T-1 channel banks on the other spans. Jeremy McNamara Anton Tinchev wrote: Need to buy 2-3 channel banks for some asterisk deployments... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best E1 channel bank?
what's the shortcoming of E1 channel banks? - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 2:36 PM Subject: Re: [Asterisk-Users] Best E1 channel bank? Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run your E-1 connection to the telco and run T-1 channel banks on the other spans. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best E1 channel bank?
price, price, price, price On Mon, 21 Jul 2003, johncn wrote: what's the shortcoming of E1 channel banks? - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 2:36 PM Subject: Re: [Asterisk-Users] Best E1 channel bank? Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run your E-1 connection to the telco and run T-1 channel banks on the other spans. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best E1 channel bank?
I bought second hand E400P for around $450. Jeremy McNamara wrote: Don't use E-1 channel banks. Pick up the new Digium card, TE410P, run your E-1 connection to the telco and run T-1 channel banks on the other spans. Jeremy McNamara Anton Tinchev wrote: Need to buy 2-3 channel banks for some asterisk deployments... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best software SIP client
Title: Message Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but is not the most user friendly and caused odd errors on the Asterisk console. What I am looking for is a software SIP client that is simple for users to use ( they don't have to understand SIP ) and that works reliably. Rgds, Stuart
RE: [Asterisk-Users] Music on hold Read error on sound device
Should I be able to see a process starting using mpg123 ? Because I don't ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst Sent: 21 July 2003 09:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Read error on sound device My musiconhold.conf is as below: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 random = quietmp3:/var/lib/asterisk/mohmp3,-z This has been copied from the working system as has the mp3 file into /var/lib/asterisk/mohmp3. The mp3 file also has the same permissions as the working system. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: 21 July 2003 01:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device Stuart Hirst wrote: When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? What does your musiconhold.conf file look like? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold Read error on sound device
Hi MOH seems to work fine for me now, however, one thing I did spot by reading the source when it wasn't :-) MOH MP3Player call mpg123 from '/usr/bin' and by default a mpg123 source install lives in '/usr/local/bin' these days. Might not be this affecting you, but try a: ln -s /usr/local/bin/mpg123 /usr/bin/mpg123 and see if it works :) Best Regards Darren. - Original Message - From: Stuart Hirst [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 11:42 AM Subject: RE: [Asterisk-Users] Music on hold Read error on sound device Should I be able to see a process starting using mpg123 ? Because I don't ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst Sent: 21 July 2003 09:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Read error on sound device My musiconhold.conf is as below: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 random = quietmp3:/var/lib/asterisk/mohmp3,-z This has been copied from the working system as has the mp3 file into /var/lib/asterisk/mohmp3. The mp3 file also has the same permissions as the working system. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: 21 July 2003 01:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device Stuart Hirst wrote: When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? What does your musiconhold.conf file look like? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Asterisk - SIP - AS5300 signalling missing on connect/clear call
Hi All, I seem to be having a problem with calls from Asterisk into the AS5300, I am sniffing the session between the AS5300 and the Asterisk server and I see the Asterisk server send a SIP INVITE and the AS5300 responds with a SIP 100 TRYING but then I do not see any more SIP signalling messages from the AS5300 once the call connects or clears on the ISDN side. Has anyone else experienced similar problems ? Finally I do a clear on the 7960 SIP phone and the call gets cleared. Calling in the opposite directions works perfectly ... Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on hold Read error on sound device
Darren, You are a diamond. That worked a treat. Thanks for taking the time to reply. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Smith Sent: 21 July 2003 12:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device Hi MOH seems to work fine for me now, however, one thing I did spot by reading the source when it wasn't :-) MOH MP3Player call mpg123 from '/usr/bin' and by default a mpg123 source install lives in '/usr/local/bin' these days. Might not be this affecting you, but try a: ln -s /usr/local/bin/mpg123 /usr/bin/mpg123 and see if it works :) Best Regards Darren. - Original Message - From: Stuart Hirst [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 11:42 AM Subject: RE: [Asterisk-Users] Music on hold Read error on sound device Should I be able to see a process starting using mpg123 ? Because I don't ! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst Sent: 21 July 2003 09:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Music on hold Read error on sound device My musiconhold.conf is as below: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 random = quietmp3:/var/lib/asterisk/mohmp3,-z This has been copied from the working system as has the mp3 file into /var/lib/asterisk/mohmp3. The mp3 file also has the same permissions as the working system. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: 21 July 2003 01:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold Read error on sound device Stuart Hirst wrote: When I put a call on hold the CLI shows moh starting but nothing is played. No errors are reported whilst starting moh. I have been trying lots of different things for hours now without success. Anyone got any pointers ? What does your musiconhold.conf file look like? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK call termination..
Hi, I am looking for call termination in the UK so that I can place calls via my internet line instead of buying more PSTN lines.. anyone know of amy providers in the UK.. somthing like nufone.net in the UK would be perfect.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones
Hi Nick, You'll probably run into quality problems making calls over the ISDN from Xten via *. We did which led us to try several other softphones which were better and worse, e.g. Pingtel was great from a quality point of view but the interface wasn't. We're using snom 100s at the moment which are working great (apart from a headset issue which may or may not be relevant). We also have one of the 4 port analogue cards from Digium with traditional POTS phones/faxes connected in to it. They work fine as well. For the future, we'll be getting snoms for all the new phones and keeping the existing analogues to the capacity of the card. We'll only be (only are) using softphones for out of office laptop use although have also set up an 0800 dial-in on * to enable authorised users to make free calls in from their home/mobile and then dial internal extensions, or external numbers on the company which really works very well. We started out with the ideal of a softphone and have retraced sharply from it. A hardware sip phone is more expensive yet provides many benefits in usability and quality. Using analogue phones is great for users (as little changes) and the quality is fine although per port it is a more expensive route than a hardware sip phone, certainly at the lower end. Maybe I'll be shot down in flames but that is certainly how it has panned out for us. All the best, Simon - Original Message - From: Nick Knight [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 10:21 AM Subject: [Asterisk-Users] Phones Hello all, I am a newbie to this list - and so far very impressed with the functionality of Asterisk. So far I have setup a simple soft phone running on a windows PC making calls to other SIP soft phones. Later this week I hope to get UK ISDN2e up and running with it! My question is I would like the experience and feedback from users about what equipment/software you are all using for phones to connect to Asterisk, so fat I have been playing with xten soft phone which works very well but before I make my decision on which phones to use would like the feedback of the group. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK call termination..
We use our own gateway for h323 and sip shortly. Contact me offline. Tan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 1:43 PM Subject: [Asterisk-Users] UK call termination.. Hi, I am looking for call termination in the UK so that I can place calls via my internet line instead of buying more PSTN lines.. anyone know of amy providers in the UK.. somthing like nufone.net in the UK would be perfect.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamically setting up/tearing down extensions
Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be web-based. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I have a test dialplan with one call to Playback() - just plays a wav file then exits. When DynExtenDB() is called, it adds one extra step that calls DynExtenDB_Free()... --If I let the wav file play to the end, DynExtenDB_Free() is called properly. If I hang up prematurely, it isn't, and it also isn't called if I set the dialplan to dial out (for example, to forward the call to my cell phone). --If DynExtenDB_Free() *is* called properly, and I then make another call, DynExtenDB() doesn't seem to be called again. --I'm not sure that setting up a dialplan for extension 'h' is a good idea. What if I call, and then someone else calls and I hang up in the middle of the call? I am ready and willing to make changes to the source to DynExtenDB. In fact, I'd like to get it to a point where it could be used in a production environment. But I have a lot of questions before I can do that. BTW, I have looked in the archives, and it's been suggested that maybe AGI is a better way to handle this sort of thing - but wouldn't the same issues still exist?? Thanks SJS -- Steven J. Sobol, Geek In Charge, JustThe.net Microsoft must think they're a navy, they open so many ports. --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Hi Steven! Small world isn't it? On Mon, 2003-07-21 at 15:52, Steven J. Sobol wrote: Hello, * newbie here, I've been lurking on the list for a few months now. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I haven't found this DynExtenDB however. Could you provide me with some pointers to it? PS: We never finished the Aegir Addon stuff. Maybe we can do that over iaxtel sometime? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Hey! On 21 Jul 2003, Armand A. Verstappen wrote: I've been lurking on the list for a few months now. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I haven't found this DynExtenDB however. Could you provide me with some pointers to it? http://andreasotto.net/asterisk/ PS: We never finished the Aegir Addon stuff. Maybe we can do that over iaxtel sometime? I haven't forgotten about it, I just haven't had time to do it. I moved about 2,500 miles across the US from Ohio to California at the end of last month. I also have to do a reinstall of my Aegir/Midgard setup since I managed to break it. I'll start on this this week. -- Steven J. Sobol, Geek In Charge, JustThe.net Microsoft must think they're a navy, they open so many ports. --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite Build 1016
Has anyone had X-Lite Build 1016 working with Asterisk and if so what settings within the X-Lite client did you use ? Rgds, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anyone with X100P Callerid working outside US ?
I'm just curious if anyone has the X100P Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
Can you have them try with just a headset? Mark On Mon, 21 Jul 2003, Linus Surguy wrote: We're currently running a PSTN - SIP gateway with Asterisk. We also run IAX/SIP - PSTN. We have performed a test where the call is routed UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... I did ask them to turn down the speakers and retest and it appears the echo was still there. I don't quite understand how the echo is being introduced in this case. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Authentication bug?
Might enter it in the new Asterisk bug tracker Mark On Mon, 21 Jul 2003, Tan Aks wrote: Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
On Tue, 22 Jul 2003, Jeremy McNamara wrote: Karl (klasstek) and myself (mainly Karl) has spent a few clock cycles figuring out how to make dynamic extensions happen, but we had no real motivation to finish the task. Well, I'd certainly be willing to pick up the project from you. I think it should be done in the core, rather than in a module, but that's just my observation, and I've only taken a very cursory look at certain parts of the Asterisk source. Find either one of us on IRC or search the mailing list archives. Yes, sir. -- Steven J. Sobol, Geek In Charge, JustThe.net Microsoft must think they're a navy, they open so many ports. --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
One can use the retrieve_extensions_from_mysql.pl script and then issue a extensions reload command to asterisk. The pending calls are unaffected and the final substitution of the new dialplan is done in a very short time. regards Martin On Tue, 22 Jul 2003, Jeremy McNamara wrote: DynExtenDB is not even close to being the proper way to achieve dynamic extensions. Karl (klasstek) and myself (mainly Karl) has spent a few clock cycles figuring out how to make dynamic extensions happen, but we had no real motivation to finish the task. Find either one of us on IRC or search the mailing list archives. Jeremy McNamara Steven J. Sobol wrote: Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be web-based. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I have a test dialplan with one call to Playback() - just plays a wav file then exits. When DynExtenDB() is called, it adds one extra step that calls DynExtenDB_Free()... --If I let the wav file play to the end, DynExtenDB_Free() is called properly. If I hang up prematurely, it isn't, and it also isn't called if I set the dialplan to dial out (for example, to forward the call to my cell phone). --If DynExtenDB_Free() *is* called properly, and I then make another call, DynExtenDB() doesn't seem to be called again. --I'm not sure that setting up a dialplan for extension 'h' is a good idea. What if I call, and then someone else calls and I hang up in the middle of the call? I am ready and willing to make changes to the source to DynExtenDB. In fact, I'd like to get it to a point where it could be used in a production environment. But I have a lot of questions before I can do that. BTW, I have looked in the archives, and it's been suggested that maybe AGI is a better way to handle this sort of thing - but wouldn't the same issues still exist?? Thanks SJS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] URGENT! brandly new Wildcard E400P for sale at $1000
A brandly new E400P 128 channel PRI, you can find more information on the Digium's site. The card was not used before, I sell it cause our company just don't need, we use Cisco AS5300. I can offer you the PCI PRI at $940 if bought in the next 2 days, I can send it to you via FedEx or DHL express. Please email me for more details! Regards, Kalin - IPPN Networks LTD http://www.ippn.net technical manager mobile: +35998804462-
[Asterisk-Users] CDR question
Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crashes when trying to load G.729module.
Try to install the new codec code that is available in ftp://ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary/codec_g729b.so place it in /usr/lib/asterisk/modules and restart asterisk (or try to start it). There is also a new command available g.729 show license usage and a few fixes to the code. Write back about the results. regards Martin On Sun, 20 Jul 2003, Anton Tinchev wrote: Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] anyone with X100P Callerid working outside US ?
How can I use ztmonitor to figure out the caller id sent by the telco? Because it is not working for me in Chicago. - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 9:03 PM Subject: Re: [Asterisk-Users] anyone with X100P Callerid working outside US ? It's possible that your telco first transmits the DID (your number) and then later on the callerid ... Did you listen for it with ztmonitor ? If my suspicion is right ? regards Martin On Mon, 21 Jul 2003, Dan wrote: Hi Martin, For me it just display my own PSTN number, extracted as caller id from the PSTN line. I am located in Romania. Best regards, Dan P.S. Some analog phones with internal caller id displays the same number, but others (especially some Siemens ones) display the correct caller id. I think that the X100P card does not extract the correct part of the callerid information. - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 8:25 PM Subject: [Asterisk-Users] anyone with X100P Callerid working outside US ? I'm just curious if anyone has the X100P Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] anyone with X100P Callerid working outsideUS ?
I don't know yet. However you should be able to hear some wierd signal that is callerid codec in FSK mode. regards Martin On Mon, 21 Jul 2003, Tamas Levente wrote: How can I use ztmonitor to figure out the caller id sent by the telco? Because it is not working for me in Chicago. - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 9:03 PM Subject: Re: [Asterisk-Users] anyone with X100P Callerid working outside US ? It's possible that your telco first transmits the DID (your number) and then later on the callerid ... Did you listen for it with ztmonitor ? If my suspicion is right ? regards Martin On Mon, 21 Jul 2003, Dan wrote: Hi Martin, For me it just display my own PSTN number, extracted as caller id from the PSTN line. I am located in Romania. Best regards, Dan P.S. Some analog phones with internal caller id displays the same number, but others (especially some Siemens ones) display the correct caller id. I think that the X100P card does not extract the correct part of the callerid information. - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 21, 2003 8:25 PM Subject: [Asterisk-Users] anyone with X100P Callerid working outside US ? I'm just curious if anyone has the X100P Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Yes, you can contact over the manager interface (you need to setup a user/pass in /etc/asterisk/manager.conf). I've sent a short perl script how to do that some time ago. Now notice that extensions reload only renews extensions without touching other modules. regards Martin On Mon, 21 Jul 2003, Steve Sobol wrote: At 02:06 PM 7/21/2003 -0500, you wrote: One can use the retrieve_extensions_from_mysql.pl script and then issue a extensions reload command to asterisk. The pending calls are unaffected and the final substitution of the new dialplan is done in a very short time. I want to explore truly dynamic extensions as a long-term project, but this might be an excellent short-term solution. Can the reload be done without being root? -- Steven J. Sobol, Geek In Charge, JustThe.net POTS: Toll Free from anywhere in the USA or Canada, 888.480.4NET (4638) HTTP: www.JustTheNetLLC.com MAIL: 5686 Davis Drive, Mentor on the Lake, OH 44060-2752 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Authentication bug?
Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan Yes. Any time I change authentication (on/off) or codec ordering, I generally restart Asterisk from scratch. This may not be required, but prior experiences taught me that lesson, and I have not tested to see if the bugs have been fixed. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs
I don't know if 911 uses caller ID or BTN (Billing Telephone Number) 900 calls, operator calls, and 800 calls use the BTN not the Caller ID... Anyone 3. Re: E911 and asterisk (Martin Pycko) Message: 3 Date: Mon, 21 Jul 2003 12:05:38 -0500 (CDT) From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E911 and asterisk Reply-To: [EMAIL PROTECTED] Isn't that enough to set up a proper Caller ID NAME ? Martin On Mon, 21 Jul 2003, Alex Lopez wrote: I have a client that would like to use asterisk to link their multiples locations together. However, if a person in the remote office dials 911, How can the 911 operator determine WHERE the emergency is?? Since all calss would be going out of the PRI in the main location, the police/fire trucks will show up at our COLO!! I know that there are some that are doing this multi site setup, how did they handle 911 services??? For now I am quoting a one port FXO card to be placed in each location, that will in turn connect to a POTS line. However, even though we can use it for the alarm system and it is a kind of insurance I would like to do away with it! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP Authentication bug?
I have seen this, along with other strange SIP auth issues. I just thought that you HAD to stop and restart * for the changes in the sip.conf file to be reread. I also have not been able to get auth to work. If I put a password in the Windopws Messenger field asterisk does not authenticate. I have tried the plaintect and insecure options in the config file. But, Nothing... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, July 21, 2003 4:44 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #873 - 16 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] Date: Mon, 21 Jul 2003 12:32:23 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Authentication bug? Reply-To: [EMAIL PROTECTED] Might enter it in the new Asterisk bug tracker Mark On Mon, 21 Jul 2003, Tan Aks wrote: Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'Echo' - I'm sure a common topic
I'll try to set up a retest and report back. Linus Can you have them try with just a headset? Mark On Mon, 21 Jul 2003, Linus Surguy wrote: We're currently running a PSTN - SIP gateway with Asterisk. We also run IAX/SIP - PSTN. We have performed a test where the call is routed UK PSTN - Digium E1 card - Asterisk GW - SIP G.711 - FWD - X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test Its not perhaps as simple as acoustic echo on the softphone side heading back to the PSTN. IE - speakers and microphone? In which case, the user needs to get a headset... I did ask them to turn down the speakers and retest and it appears the echo was still there. I don't quite understand how the echo is being introduced in this case. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF crashes chan_capi
Quoting Myself :( Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set softdtmf=1 in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection? The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP client is SJPhone (G711) on Windows connected to * via a 100Mbit switched LAN. PSTN connection is using latest chan_capi on an AVM B1 PCI v4 card (a steal at £1.20 on ebay :)). I can call the * box using the SIP client and interact with the voicemail app with no problems using in-band DTMF. I can also call in from the PSTN through the capi interface and interact with the IVR menu with no problems. Finally I can bridge the CAPI and SIP channels and hear DTMF digits entered on the PSTN phone with no problems (they are also detected and displayed on the console). However when the CAPI and SIP channels are bridged, entering more than a couple of DTMF digits into the _SIP_ client appears to crash the channel: neither party gets disconnected, but there is no longer any audio in either direction and new calls (inbound or outbound) trying to use the CAPI channel fail. Once locked if I enter capi info in the * console it return nothing and trying to autocomplete capi commands e.g. capi [TAB] just locks the console up. Entering capiinfo and lsmod at the command prompt suggests the driver is ok. The only way of getting it working again is to restart *. When I switch to softdtmf, everything seems to work fine, but I noticed that even though DTMF signalling works fine on the IVR menu, once the call is bridged DTMF digits entered on the PSTN phone are not displayed on the console like before. Further to my earlier post... I've replicated this problem using a passive Fritz PCI card, so my bargain B1 card appears not to be the problem. I can only replicate the lock ups when bridging a call directly from chan_capi to chan_sip (or vica versa) and I use in-band DTMF detection in the SIP channel. I have tried various other combinations, including SIPSIP, CAPICAPI, CAPIMEETMECAPI and CAPIMEETMESIP, and they work without a problem. The SIP client I am using is SJPhone (which does only in-band DTMF), and if I switch off the in-band detection in sip.conf it works fine, but of course I can no longer signal to * from the SIP phone. I can't test whether out of band signalling is ok as SJPhone doesn't support this. I find it hard to believe that this is a capi problem - I would have expected see some other capi related problems aside from this single (easily repeatable) error. Jamie Neil Versado I.T. Services Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Robbed bit signalling debugging
I'm trying to debug a problem with robbed bit signalling on a T1 coming into an Asterisk box on a T100P card. Specifically, I need to look at the signalling timing. Is there a way to turn on this kind of debugging in Asterisk, similar to what 'pri debug' does? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] anyone with X100P Callerid working outsideUS ?
On Mon, 2003-07-21 at 19:25, Martin Pycko wrote: I'm just curious if anyone has the X100P Callerid receiving working outside US. It does not work in the Netherlands. The Netherlands does not use FSK signalling, but DTMF signalling: 1) polarity reversal 2) DTMF: DNumberC 3) ring signal where D and C are the DTMF tones 'D' and 'C' respectively, signalling start and end of DTMF Caller-ID transfer. Exact specification (including length of tones and pauses) is in this document: http://www.kpn.com/common/downloads/01_Part2-PSTN_V32.pdf paragraph 6.2.3. At least Sweden and Denmark use very similar CLIP protocols, the difference being mainly in the start and end tones used. The different protocol also bites on the other end, as asterisk will send callerid information to a phone connected to a TDM40B for example using the FSK protocol. Dutch phones don't understand FSK, and hence don't pick up on the caller id. I'm very interested in solving this problem, as it makes asterisk only usable in the Netherlands using ISDN BRI or PRI on the PSTN side, and imported phones on the (analog) internal side. I just don't have any idea where in the source, and how... One possible solution for the 'inside' problem: There's one company in the Netherlands offering telefony over cable infrastructure, they use FSK signalling for Caller-ID presentation. I'll get my hands on a phone suited for their network soon, wich will allow me to verify if they work with asterisk. -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
[Asterisk-Users] PAnasonic And Asterisk
Dear Pals One customer has a Panasonic PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now, This customer has 10 wireless links to his branches, wireless working great now, no voice at the present MY IDEA : T1Card into the Panasonic (additional to the E1) connected to a T1 Digium Card into Asterisk as far as I know the T1 Card can be configured as EM to act as extensions. , the question do you think the configuration of the T1 Card is almost the same as T1? (in Mexico we do not have T1´s only E1´s and our Panasonic knowledge is for E1) or any idea how to do the following needs have 20 ATA´s connected into the Wireless network to the asterisk and the 20 ports can be extensions for the Panasonic? Can I use G.723.1 in this configuration? Someone has a better idea? Regards Humberto Atristain
[Asterisk-Users] Using asterisk for a 911 call center....
Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? Gene Kochanowsky Solution Sciences, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using asterisk for a 911 call center....
911 trunks are usually delivered to public-safety answering points (PSAP) on analog reverse-battery facilities. (The PSAP provides battery toward the CO). ANI is provided using MF tones. The PSAP equipment must take the ANI and use it to submit a database query to lookup the caller's address (ALI - automatic location identification) to an external database that is usually operated by the ILEC or its contractor. The ANI and ALI information must be presented to the 911 call taker immediately upon answering the call. There is definitely a place for Asterisk in the public-safety telecomm field, but a lot of work would be needed for it to handle calls in a PSAP. If you're interested in learning more about PSAP equipment, take a look at the web pages for Plant Equipment, Inc, Positron, Inc. and Zetron, Inc. On Mon, 21 Jul 2003, Gene Kochanowsky wrote: Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using asterisk for a 911 call center....
Daryl, thanks for the info! I'll checkout those companies. Gene -Original Message- From: Daryl Jones [mailto:[EMAIL PROTECTED] Sent: Monday, July 21, 2003 11:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Using asterisk for a 911 call center 911 trunks are usually delivered to public-safety answering points (PSAP) on analog reverse-battery facilities. (The PSAP provides battery toward the CO). ANI is provided using MF tones. The PSAP equipment must take the ANI and use it to submit a database query to lookup the caller's address (ALI - automatic location identification) to an external database that is usually operated by the ILEC or its contractor. The ANI and ALI information must be presented to the 911 call taker immediately upon answering the call. There is definitely a place for Asterisk in the public-safety telecomm field, but a lot of work would be needed for it to handle calls in a PSAP. If you're interested in learning more about PSAP equipment, take a look at the web pages for Plant Equipment, Inc, Positron, Inc. and Zetron, Inc. On Mon, 21 Jul 2003, Gene Kochanowsky wrote: Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MYSQL Table Structure
For the information of others (and Google) this is the table structure I used to get cdr_mysql working: CREATE TABLE cdr ( calldate varchar(255) NOT NULL default '', clid varchar(255) NOT NULL default '', src varchar(255) NOT NULL default '', dst varchar(255) NOT NULL default '', dcontext varchar(255) NOT NULL default '', channel varchar(255) NOT NULL default '', dstchannel varchar(255) NOT NULL default '', lastapp varchar(255) NOT NULL default '', lastdata varchar(255) NOT NULL default '', duration int(11) NOT NULL default 0, billsec int(11) NOT NULL default 0, disposition int(11) NOT NULL default 0, amaflags int(11) NOT NULL default 0, accountcode varchar(255) NOT NULL default '', KEY src (src), KEY clid (clid), KEY dst (dst), KEY accountcode (accountcode)) TYPE=MyISAM COMMENT='Asterisk CDR table';
[Asterisk-Users] g729 + oh323
Hello, Is Oh323 supports g729 codec from digium? I saw an g729 option in the oh323.conf but I have also read some post in the mailing list saying that oh323 doesn't support g729 codec from digium. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MYSQL Table Structure
Cant have been that obvious... - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 22, 2003 4:32 PM Subject: Re: [Asterisk-Users] MYSQL Table Structure On Monday 21 July 2003 23:21, Aaron Martin wrote: For the information of others (and Google) this is the table structure I used to get cdr_mysql working: CREATE TABLE cdr ( calldate varchar(255) NOT NULL default '', clid varchar(255) NOT NULL default '', src varchar(255) NOT NULL default '', dst varchar(255) NOT NULL default '', dcontext varchar(255) NOT NULL default '', channel varchar(255) NOT NULL default '', dstchannel varchar(255) NOT NULL default '', lastapp varchar(255) NOT NULL default '', lastdata varchar(255) NOT NULL default '', duration int(11) NOT NULL default 0, billsec int(11) NOT NULL default 0, disposition int(11) NOT NULL default 0, amaflags int(11) NOT NULL default 0, accountcode varchar(255) NOT NULL default '', KEY src (src), KEY clid (clid), KEY dst (dst), KEY accountcode (accountcode) ) TYPE=MyISAM COMMENT='Asterisk CDR table'; Captain Obvious says: [EMAIL PROTECTED]:/cvs/asterisk# head -11 doc/cdr_mysql.txt | tail -1 CREATE TABLE cdr ( [EMAIL PROTECTED]:/cvs/asterisk# grep cdr_mysql.txt doc/CVS/Entries /cdr_mysql.txt/1.1.1.1/Wed Feb 12 13:59:15 2003// -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
