Re: [asterisk-users] time on asterisk
Hi Sir, I tried restarting asterisk, but still it has the wrong time. I tried restarting the system, then start asterisk it still uses the wrong time. I also tried recompiling asterisk, checked i have the correct time on the system,nbsp; then restart the system then start asterisk but still i get the wrong time. My system time (currently) Thu Jun 12 15:12:11 GST 2008 on asterisk i use EPOCH to look at the time, nbsp; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) i would really appreciate any help. TIA ron --- On Thu, 6/12/08, Tilghman Lesher lt;[EMAIL PROTECTED]gt; wrote: From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 1:42 AM On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; between the asterisk time and the system time. could it be because i gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; of timezone on the system? How can I fix this prob? Yes, that's probably the reason. The system timezone is cached once at startup, for performance reasons. The only way to get it to pick up the new timezone is a restart. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 phones, BRI and Analogue cards
Hi Hans, Can't you leave the picking up of the cli to the isdn line? Even if it is an ISDN1 (just a B-channel and a D-channel), the chances of tranferring channel info, like CLI, is better. If a call comes in over the POTS line, then I still need to get CLI over it. I'm not sure if the ISDN can be specified to replace the POTS analogue line, whilst retaining the analogue line + ADSL. Cheers, Ade. Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.524 / Virus Database: 269.24.6 - Release Date: 03/06/2008 00:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring QoS
Hello Fellow Users, I am looking for a way - using certain software or other techniques - to monitor, measure, and improve the quality of service for Asterisk system. During the last while, it seems the quality has decreased and am trying to look for ways to get things going well again. Thanks, Murdock ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to turn on the H323 logging on Asterisk
Hi Tony, Thanks a lot for the tips. I have turned on the logging and saw them in the console. However, this applies only for startup, when I try to register a user, which I cannot succeed, there is no logging done. Do you think you can give me an idea why my user cannot register? All I want to do is, send the register to a GK which has the IWF functionality. Attached I am sending my extensions.conf and ooh323.conf files. Sema is the end user and I believe mypeer is the GK definition, right? [mypeer1] type=peer ;context=context2 ip=10.192.192.6 ; UPDATE with appropriate ip address port=1720; UPDATE with appropriate port allow=all ;e164=101 [Sema] type=friend context=default ip=10.192.192.36 ; UPDATE with appropriate ip address port=1820; UPDATE with appropriate port ;disallow=all allow=all e164=05336887755 rtptimeout=60 dtmfmode=rfc2833 I would really appreciate if you can take a look. Kr, Sema On Wed, Jun 11, 2008 at 1:25 PM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jun 11, 2008 at 10:40:41AM +0300, Sema Arca wrote: Hi, Does anybody know how I can turn on the logging for H323 in Asterisk? I have set the logging path and the file name in the ooh323.conf file however it did not help. The file is created but is empty. I want to, if possible, turn on the logging in DEBUG level. ooh323 does not have debug-to-file. You enable debugging with ooh323 debug, and then the debug information is sent to the verbose channel, which normally goes to the console and may go to one of the general log files, according to the settings in logger.conf. ooh323 debugging is stopped by giving ooh323 no debug. The file name is ooh323c.conf (note the extra 'c'). No, ooh323.conf is correct. The 'c' is used in the name of the stack, but not in the name of the Asterisk channel or the conf file. It is used by chan_ooh323c, rather than chan_h323. chan_ooh323c is unmaintained and not recommended for new installations. This was because until recently, the most up-to-date chan_ooh323 driver and stack were the ones in the 1.2 branch of asterisk-addons. However, I recently ported the 1.2 version forward to 1.4, trunk and 1.6.0, and added a couple of bug fixes. Those changes were accepted into SVN, so that all those variants are now up to date. It should therefore now be easy to keep them maintained as far as Asterisk API changes are concerned. Having tried chan_h323, chan_oh323 and chan_ooh323, I *would* strongly recommend chan_ooh323 over the first two. It is clean and lightweight, uses the Asterisk RTP stack (and can therefore bridge properly), and doesn't creak under the bloat of OpenH323 like the first two do. I don't know whether Objective Systems have abandoned chan_ooh323 and the ooh323c stack, but it would be great to see them moved from -addons into the main Asterisk tree. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ooh323.conf Description: Binary data extensions.conf Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
Nhadie Ramos wrote: Hi Sir, I tried restarting asterisk, but still it has the wrong time. I tried restarting the system, then start asterisk it still uses the wrong time. I also tried recompiling asterisk, checked i have the correct time on the system, then restart the system then start asterisk but still i get the wrong time. My system time (currently) Thu Jun 12 15:12:11 GST 2008 on asterisk i use EPOCH to look at the time, NoOp(SIP/105101-00857e60, DATE: 20080612-081147) i would really appreciate any help. TIA ron --- On *Thu, 6/12/08, Tilghman Lesher /[EMAIL PROTECTED]/* wrote: From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 12, 2008, 1:42 AM On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: I'm using gotoiftime on asterisk, but it seemsnbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Yes, that's probably the reason. The system timezone is cached once at startup, for performance reasons. The only way to get it to pick up the new timezone is a restart. -- Tilghman Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. My situation seems unique because I am not using a router even at this point. I was given a sheet of ip addresses and was told just to provision by devices with the given ip's and they would handle the rest. My devices are hooked directly to their switch in my location. This hasn't been an issue up until now because I only had analog (mediatrix and audiocodes 24 port gateways x 4) connected to the switch. Now I am going to a software based dialer (i.e. asterisk/ vicidial) and have run into these problems. Thanks again, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Wednesday, June 11, 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber On Wed, 11 Jun 2008, Mark Adams wrote: (I know there are security issues as they have been additional users created on my server and irc junk was put in the home folder) If the box has been compromised, the only recourse is to erase the drives and start over. You can't trust anything on the box. Off the top of my head, this is how I would approach the problem. 1) Identify how the box was compromised. (A client box was recently (last 30 days) hacked. It was an old AAH installed by the client. The hacker used the default password on the admin account to exploit a buffer overflow in crond to gain root.) 2) Save any essential data -- and only the data, no executables. 3) Take the box off the Internet. 4) Boot DBAN and let it do it's thing. 5) Install a minimal OS from CD/DVD. 6) Clean up after the install -- turn off services, delete users, delete packages, add packages, etc. 7) Bring up to current patch level from your private repository. 8) Expose the box to the Internet. 9) Cross your fingers and actively monitor the box. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 phones, BRI and Analogue cards
Hi; I would like just to know one thing: Where did u find a good IAX IP Phone? I am looking in the market since long time to buy such device and did not find a reliable one till now. Any advise? Regards Bilal --- Hi, I've been asked to spec up a small Asterisk system, which needs to: - Connect to ISDN2e (I'm thinking of using a B100P card here) - Connect to the POTS (A400P with 1 FXO) - Allow remote phones (thinking of an ETC 6050 utilising IAX2) It is a requirement that the POTS analogue card picks up CLI information - and I'm in the UK which, historically, has lousy CLI support certainly, my AX100P doesn't do it... does anyone have any good news about the A400P, or do I need to be hunting down a genuine Digium card? I'm further assuming that an IAX2 phone will work far more reliably through firewalls non-static IP addresses (Asterisk box will be on a static IP, remote/roaming office may not be) than a SIP phone, based on my experiences of getting IAX2 between Asterisks to work. So -- am I on the right lines with the hardware I've specced above, or should I be looking at alternatives? Can't you leave the picking up of the cli to the isdn line? Even if it is an ISDN1 (just a B-channel and a D-channel), the chances of tranferring channel info, like CLI, is better. I would leave the pots-interfaces for the people stuck with an ordinary phone (or fax).. Did you consider sipura 3102? Easier to scale than analogue cards. And perhaps easier to deploy, no spof,.. hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Command Option D Early Bridged
Dear All, The documentation of the Dial Command, says the following about Option D: D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. However, in my experience, the timing the call get bridged is not consistance, sometimes even before sending the DTMF strings. Anyone share this experience? How to make sure that the call only get bridged after sending the DTMF strings. Regards, TC Important Warning! *** This electronic communication (including any attached files) may contain confidential and/or legally privileged information and is only intended for the use of the person to whom it is addressed. If you are not the intended recipient, you do not have permission to read, use, disseminate, distribute, copy or retain any part of this communication or its attachments in any form. If this e-mail was sent to you by mistake, please take the time to notify the sender so that they can identify the problem and avoid any more mistakes in sending e-mail to you. The unauthorised use of information contained in this communication or its attachments may result in legal action against any person who uses it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to turn on the H323 logging on Asterisk
I am still looking to know if all of these h323's are able to work as gatekeeper, so endpoint can register? About chan_ooh323 and using It is clean the Asterisk RTP stack (and can therefore bridge properly), and doesn't creak under the bloat of OpenH323 like the first two do: The other two: how they use the RTP stack if they do not use Asterisk RTP? And what do u mean by bridge properly? (How?) Your kindly help is high appreciated. Regards Bilal --- In article [EMAIL PROTECTED], Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jun 11, 2008 at 10:40:41AM +0300, Sema Arca wrote: Hi, Does anybody know how I can turn on the logging for H323 in Asterisk? I have set the logging path and the file name in the ooh323.conf file however it did not help. The file is created but is empty. I want to, if possible, turn on the logging in DEBUG level. ooh323 does not have debug-to-file. You enable debugging with ooh323 debug, and then the debug information is sent to the verbose channel, which normally goes to the console and may go to one of the general log files, according to the settings in logger.conf. ooh323 debugging is stopped by giving ooh323 no debug. The file name is ooh323c.conf (note the extra 'c'). No, ooh323.conf is correct. The 'c' is used in the name of the stack, but not in the name of the Asterisk channel or the conf file. It is used by chan_ooh323c, rather than chan_h323. chan_ooh323c is unmaintained and not recommended for new installations. This was because until recently, the most up-to-date chan_ooh323 driver and stack were the ones in the 1.2 branch of asterisk-addons. However, I recently ported the 1.2 version forward to 1.4, trunk and 1.6.0, and added a couple of bug fixes. Those changes were accepted into SVN, so that all those variants are now up to date. It should therefore now be easy to keep them maintained as far as Asterisk API changes are concerned. Having tried chan_h323, chan_oh323 and chan_ooh323, I *would* strongly recommend chan_ooh323 over the first two. It is clean and lightweight, uses the Asterisk RTP stack (and can therefore bridge properly), and doesn't creak under the bloat of OpenH323 like the first two do. I don't know whether Objective Systems have abandoned chan_ooh323 and the ooh323c stack, but it would be great to see them moved from -addons into the main Asterisk tree. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Data Calls
Tilghman Lesher schrieb: On Wednesday 11 June 2008 10:20:15 Brent Davidson wrote: There is not, although I don't see any reason why it couldn't be done. There is a ZapRAS application which performs much of this same function, although it only works on ISDN lines (where the line signal is already a digital stream of bits). But ZapRAS can only be used to dial-in with another ISDN Modem on my side, right. If i have a simple analouge modem there will be no data connection because of the different protocolls. Is this correct? Regards, Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 lt;[EMAIL PROTECTED]gt; wrote: From: mkn0014 lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: gt; Hi Sir, gt; gt; I tried restarting asterisk, but still it has the wrong time. gt; gt; I tried restarting the system, then start asterisk it still uses the gt; wrong time. gt; gt; I also tried recompiling asterisk, checked i have the correct time on gt; the system, then restart the system then start asterisk but still i gt; get the wrong time. gt; gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008 gt; gt; on asterisk i use EPOCH to look at the time, gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) gt; gt; i would really appreciate any help. TIA gt; gt; ron gt; gt; --- On *Thu, 6/12/08, Tilghman Lesher gt; /lt;[EMAIL PROTECTED]gt;/* wrote: gt; gt; From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; gt; Subject: Re: [asterisk-users] time on asterisk gt; To: Asterisk Users Mailing List - Non-Commercial Discussion gt; lt;asterisk-users@lists.digium.comgt; gt; Date: Thursday, June 12, 2008, 1:42 AM gt; gt; On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; gt; between the asterisk time and the system time. could it be because i gt; gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; gt; of timezone on the system? How can I fix this prob? gt; gt; Yes, that's probably the reason. The system timezone is cached once at gt; startup, for performance reasons. The only way to get it to pick up the new gt; timezone is a restart. gt; gt; -- gt; Tilghman gt; gt; Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 phones, BRI and Analogue cards
bilal ghayyad wrote: I would like just to know one thing: Where did u find a good IAX IP Phone? I am looking in the market since long time to buy such device and did not find a reliable one till now. Any advise? I haven't tried any yet; but http://x100p.eu have a few for sale; plus there are some on eBay, one of which I intend to try out, as it looks very similar (identical) to the 6050 for some £30 less... Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.3.0/1498 - Release Date: 11/06/2008 19:13 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday the 13th lucky asterisk appliance day
Hi,
We can always count on Dean Collins to arrange interesting things and
this Friday, June 13th, we're LUCKY to have a very interesting offer
if you happen to be looking for an applicane.
See http://VoipUsersConference.org
This Friday the 13th we'll be hearing about the newest Asterisk
appliance on the block the Vdex-40 from Technoco.
TAA.com are the USA distributors and Dean Collins is helping them
launch their efforts with their USA/Canada sales push.
Participants on Fridays call will be given a 7 day time limited url
where they can purchase a single 'demo' appliance at 40% off rrp.
Dial in this Friday to find out more.
IRC.Freenode.net #voip-users-conference
PSTN;: Call (724) 444-7444 and enter 22622# 1#
Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#)) ; use #1 is you
do not join Talkshoe
TS.x2z.eu resolves to the above IP
http://food4wine.ning.com has news, forums, blogs, etc
RSS http://feeds.feedburner.com/AstUser
Trademarks are copyright their various owners.
___
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Re: [asterisk-users] How to turn on the H323 logging on Asterisk
In article [EMAIL PROTECTED], bilal ghayyad [EMAIL PROTECTED] wrote: I am still looking to know if all of these h323's are able to work as gatekeeper, so endpoint can register? I think they all run only as a gateway, not a gatekeeper, but I'm not 100% certain. About chan_ooh323 and using It is clean the Asterisk RTP stack (and can therefore bridge properly), and doesn't creak under the bloat of OpenH323 like the first two do: The other two: how they use the RTP stack if they do not use Asterisk RTP? Looking at http://www.voip-info.org/wiki/index.php?page=Asterisk+H323+channels it appears I was only partially correct. I never got chan_h323 working, so have less experience of that. According to the above wiki page, chan_h323 does use the Asterisk RTP stack, although it still uses the OpenH323 library for the protocol part. I used chan_oh323 for a long time 2 or 3 years ago, and it definitely didn't use Asterisk's RTP stack, nor its codecs. It used the ones that are part of OpenH323, and communicated with the chan_oh323 driver using pipes (I guess in slin format). It was very profligate in its use of system resources (file descriptors, CPU, etc), such that even on a dual Xeon system we got degradation above about 15 simultaneous calls. With both those versions, you easily run into version number hell with OpenH323 and PWlib, which is another reason, IMHO, to avoid them. I tried chan_ooh323 much more recently, and it just felt cleaner, more streamlined and better integrated with Asterisk. However, I have not yet used it in production. When I need to, this is the H.323 driver that I will use, and if necessary bug fix and/or enhance. And what do u mean by bridge properly? (How?) I guess since chan_h323 does indeed use Asterisk RTP, that it can bridge channels at the RTP packet level just like chan_ooh323 and chan_sip can. But chan_oh323 always had to pass audio through the Asterisk core because it didn't use Asterisk RTP. Your kindly help is high appreciated. Regards Bilal Cheers Tony --- In article [EMAIL PROTECTED], Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jun 11, 2008 at 10:40:41AM +0300, Sema Arca wrote: Hi, Does anybody know how I can turn on the logging for H323 in Asterisk? I have set the logging path and the file name in the ooh323.conf file however it did not help. The file is created but is empty. I want to, if possible, turn on the logging in DEBUG level. ooh323 does not have debug-to-file. You enable debugging with ooh323 debug, and then the debug information is sent to the verbose channel, which normally goes to the console and may go to one of the general log files, according to the settings in logger.conf. ooh323 debugging is stopped by giving ooh323 no debug. The file name is ooh323c.conf (note the extra 'c'). No, ooh323.conf is correct. The 'c' is used in the name of the stack, but not in the name of the Asterisk channel or the conf file. It is used by chan_ooh323c, rather than chan_h323. chan_ooh323c is unmaintained and not recommended for new installations. This was because until recently, the most up-to-date chan_ooh323 driver and stack were the ones in the 1.2 branch of asterisk-addons. However, I recently ported the 1.2 version forward to 1.4, trunk and 1.6.0, and added a couple of bug fixes. Those changes were accepted into SVN, so that all those variants are now up to date. It should therefore now be easy to keep them maintained as far as Asterisk API changes are concerned. Having tried chan_h323, chan_oh323 and chan_ooh323, I *would* strongly recommend chan_ooh323 over the first two. It is clean and lightweight, uses the Asterisk RTP stack (and can therefore bridge properly), and doesn't creak under the bloat of OpenH323 like the first two do. I don't know whether Objective Systems have abandoned chan_ooh323 and the ooh323c stack, but it would be great to see them moved from -addons into the main Asterisk tree. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
GMT timezone does not have daylight savings, so probably this is why you have the wrong time Select a timezone for a city and usually the correct daylight parameters are used Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com http://www.digital-opsis.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Thursday, June 12, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] time on asterisk hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 [EMAIL PROTECTED] wrote: From: mkn0014 [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: Hi Sir, I tried restarting asterisk, but still it has the wrong time. I tried restarting the system, then start asterisk it still uses the wrong time. I also tried recompiling asterisk, checked i have the correct time on the system, then restart the system then start asterisk but still i get the wrong time. My system time (currently) Thu Jun 12 15:12:11 GST 2008 on asterisk i use EPOCH to look at the time, NoOp(SIP/105101-00857e60, DATE: 20080612-081147) i would really appreciate any help. TIA ron --- On *Thu, 6/12/08, Tilghman Lesher /[EMAIL PROTECTED]/* wrote: From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, June 12, 2008, 1:42 AM On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: I'm using gotoiftime on asterisk, but it seemsnbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Yes, that's probably the reason. The system timezone is cached once at startup, for performance reasons. The only way to get it to pick up the new timezone is a restart. -- Tilghman Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
On Thu, Jun 12, 2008 at 01:59:51AM -0700, Nhadie Ramos wrote: hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. You should actually be using Asia/Singapure rather than guess. i'm not using ntp, coz when i do i also don't get the correct time. That's because you have an incorrect timezone set. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial command and its g option
I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
What services do you need exposed to the internet and on what machines? Does the fiber just terminate into your switch then? What type of switch? Can you get access to the switch? If so you can probably create access control lists. You could put your own router in front to act as a firewall or/and NAT and add your own ACLs. As already suggested, turn off all unused services. Do not use some all in one rolled up ISO such as Trixbox. Change your ssh port. If at all possible, use OpenVPN (or whatever VPN) to connect all the machines together, as well as trusted clients then block all traffic in your ACLs (or firewall) except VPN, NTP, DNS, HTTP, and whatever I am missing. BTW I am no security expert. I had a box compromised exactly as you described but the IRC junk was pegging the CPU, not Asterisk. Thanks, Steve On Thu, Jun 12, 2008 at 4:23 AM, Mark Adams [EMAIL PROTECTED] wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. My situation seems unique because I am not using a router even at this point. I was given a sheet of ip addresses and was told just to provision by devices with the given ip's and they would handle the rest. My devices are hooked directly to their switch in my location. This hasn't been an issue up until now because I only had analog (mediatrix and audiocodes 24 port gateways x 4) connected to the switch. Now I am going to a software based dialer (i.e. asterisk/ vicidial) and have run into these problems. Thanks again, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Wednesday, June 11, 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber On Wed, 11 Jun 2008, Mark Adams wrote: (I know there are security issues as they have been additional users created on my server and irc junk was put in the home folder) If the box has been compromised, the only recourse is to erase the drives and start over. You can't trust anything on the box. Off the top of my head, this is how I would approach the problem. 1) Identify how the box was compromised. (A client box was recently (last 30 days) hacked. It was an old AAH installed by the client. The hacker used the default password on the admin account to exploit a buffer overflow in crond to gain root.) 2) Save any essential data -- and only the data, no executables. 3) Take the box off the Internet. 4) Boot DBAN and let it do it's thing. 5) Install a minimal OS from CD/DVD. 6) Clean up after the install -- turn off services, delete users, delete packages, add packages, etc. 7) Bring up to current patch level from your private repository. 8) Expose the box to the Internet. 9) Cross your fingers and actively monitor the box. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
hi sir, i forgot to mention it was originally at Asia/Singapore, when i noticed that asterisk has a wrong time, that's why i tried GMT instead. regards, Ron Tzafrir Cohen wrote: On Thu, Jun 12, 2008 at 01:59:51AM -0700, Nhadie Ramos wrote: hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. You should actually be using Asia/Singapure rather than guess. i'm not using ntp, coz when i do i also don't get the correct time. That's because you have an incorrect timezone set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Thanks for the response. I have a tellabs 8813 switch provided from time warner. No I currently do not have access to the switch. I am in the process of converting from analog based dialers using dialogic hardware TO asterisk/ vicidial systems I am strictly placing sip calls to my termination provider. I do not use the linux box for anything else. This fiber connection is dedicated to sip g729 calls entirely. Yes the fiber terminates directly to the switch. There are 6 analog to voip gateways (audiocodes and mediatrix) and 1 asterisk server. The gateways and 1 asterisk server are connected to the tellabs switch, security was never an issue because for the last 2 years we only connected analog to voip gateways to the open fiber connection. Now we want to get out of the dialogic junk and replace those systems with asterisk servers. Security has become troublesome while testing the first 50-80 channel server we have. Our asterisk server has fedora 8, x windows, asterisk 1.4 I believe. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 6:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber What services do you need exposed to the internet and on what machines? Does the fiber just terminate into your switch then? What type of switch? Can you get access to the switch? If so you can probably create access control lists. You could put your own router in front to act as a firewall or/and NAT and add your own ACLs. As already suggested, turn off all unused services. Do not use some all in one rolled up ISO such as Trixbox. Change your ssh port. If at all possible, use OpenVPN (or whatever VPN) to connect all the machines together, as well as trusted clients then block all traffic in your ACLs (or firewall) except VPN, NTP, DNS, HTTP, and whatever I am missing. BTW I am no security expert. I had a box compromised exactly as you described but the IRC junk was pegging the CPU, not Asterisk. Thanks, Steve On Thu, Jun 12, 2008 at 4:23 AM, Mark Adams [EMAIL PROTECTED] wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. My situation seems unique because I am not using a router even at this point. I was given a sheet of ip addresses and was told just to provision by devices with the given ip's and they would handle the rest. My devices are hooked directly to their switch in my location. This hasn't been an issue up until now because I only had analog (mediatrix and audiocodes 24 port gateways x 4) connected to the switch. Now I am going to a software based dialer (i.e. asterisk/ vicidial) and have run into these problems. Thanks again, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Wednesday, June 11, 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber On Wed, 11 Jun 2008, Mark Adams wrote: (I know there are security issues as they have been additional users created on my server and irc junk was put in the home folder) If the box has been compromised, the only recourse is to erase the drives and start over. You can't trust anything on the box. Off the top of my head, this is how I would approach the problem. 1) Identify how the box was compromised. (A client box was recently (last 30 days) hacked. It was an old AAH installed by the client. The hacker used the default password on the admin account to exploit a buffer overflow in crond to gain root.) 2) Save any essential data -- and only the data, no executables. 3) Take the box off the Internet. 4) Boot DBAN and let it do it's thing. 5) Install a minimal OS from CD/DVD. 6) Clean up after the install -- turn off services, delete users, delete packages, add packages, etc. 7) Bring up to current patch level from your private repository. 8) Expose the box to the Internet. 9) Cross your fingers and actively monitor the box. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Then I would think IPtables should work just fine for you. You have local access to the * box? Even a simple NAT should probably work OK with a little config tweaking. Have a look here http://swik.net/iptables+sip Thanks, Steve On Thu, Jun 12, 2008 at 7:03 AM, Mark Adams [EMAIL PROTECTED] wrote: Thanks for the response. I have a tellabs 8813 switch provided from time warner. No I currently do not have access to the switch. I am in the process of converting from analog based dialers using dialogic hardware TO asterisk/ vicidial systems I am strictly placing sip calls to my termination provider. I do not use the linux box for anything else. This fiber connection is dedicated to sip g729 calls entirely. Yes the fiber terminates directly to the switch. There are 6 analog to voip gateways (audiocodes and mediatrix) and 1 asterisk server. The gateways and 1 asterisk server are connected to the tellabs switch, security was never an issue because for the last 2 years we only connected analog to voip gateways to the open fiber connection. Now we want to get out of the dialogic junk and replace those systems with asterisk servers. Security has become troublesome while testing the first 50-80 channel server we have. Our asterisk server has fedora 8, x windows, asterisk 1.4 I believe. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 6:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber What services do you need exposed to the internet and on what machines? Does the fiber just terminate into your switch then? What type of switch? Can you get access to the switch? If so you can probably create access control lists. You could put your own router in front to act as a firewall or/and NAT and add your own ACLs. As already suggested, turn off all unused services. Do not use some all in one rolled up ISO such as Trixbox. Change your ssh port. If at all possible, use OpenVPN (or whatever VPN) to connect all the machines together, as well as trusted clients then block all traffic in your ACLs (or firewall) except VPN, NTP, DNS, HTTP, and whatever I am missing. BTW I am no security expert. I had a box compromised exactly as you described but the IRC junk was pegging the CPU, not Asterisk. Thanks, Steve On Thu, Jun 12, 2008 at 4:23 AM, Mark Adams [EMAIL PROTECTED] wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. My situation seems unique because I am not using a router even at this point. I was given a sheet of ip addresses and was told just to provision by devices with the given ip's and they would handle the rest. My devices are hooked directly to their switch in my location. This hasn't been an issue up until now because I only had analog (mediatrix and audiocodes 24 port gateways x 4) connected to the switch. Now I am going to a software based dialer (i.e. asterisk/ vicidial) and have run into these problems. Thanks again, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Wednesday, June 11, 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber On Wed, 11 Jun 2008, Mark Adams wrote: (I know there are security issues as they have been additional users created on my server and irc junk was put in the home folder) If the box has been compromised, the only recourse is to erase the drives and start over. You can't trust anything on the box. Off the top of my head, this is how I would approach the problem. 1) Identify how the box was compromised. (A client box was recently (last 30 days) hacked. It was an old AAH installed by the client. The hacker used the default password on the admin account to exploit a buffer overflow in crond to gain root.) 2) Save any essential data -- and only the data, no executables. 3) Take the box off the Internet. 4) Boot DBAN and let it do it's thing. 5) Install a minimal OS from CD/DVD. 6) Clean up after the install -- turn off services, delete users, delete packages, add packages, etc. 7) Bring up to current patch level from your private repository. 8) Expose the box to the Internet. 9) Cross your fingers and actively monitor the box. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline
[asterisk-users] AGI after Hangup
Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing vs forward - was RE: Asterisk : using setvar with IP Realtime and variable inheritance
See, to get back to your answer, this is what I`m not understanding:
Again, this works fine. The problem is when I forward my calls to
another
outside line (using Polyocm phones), and need to know the ${did} value
at
that point. It's empty.
Right, so the call path is:
Provider -- Asterisk -- Polycom -- Asterisk--Provider
The problem is that Polycom is in the call path. It doesn't know anything
about Asterisk variables and so it doesn't pass them on [...].
I understand the Polycom doesn't know anything about my diaplan. BUT, the
Polycom, when doing it's forwading, must be dialing by using a specific
context, taking from a line registration. That context is taken from the
sip_registration table (in my case), context column. This is what it does
when I dial out, AND this is what it also seems to do when I forward. After
all, a forward is just an automated outgoing call...
So, a normal outgoing call out of my Polycom is using the exact same context
that a forward is using. At least, that`s my observation and my premise.
(by all means, if I am wrong somebody tell me know in what way).
What is NOT the same is that the setvar variable (did=551234) is taken
into account when dialing out, but not with a forward; it's empty.
Why? Is this WAD or a bug, or am I missing an obscure option in Asterisk?
Mick
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[asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
Hi all, I have setup an asterisk system which: 1. recieves incoming sip calls 2. ask the caller the number they want to dial, and then dial that number 3. after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. 4. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial command and its g option
just add as many extensions as you want under the Dial command extension keeping the extension number same: exten = s,n,Dial(SIP/100,100,Ttg) exten = s,n,Application here On Thu, Jun 12, 2008 at 3:25 PM, voip crazy [EMAIL PROTECTED] wrote: I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial command and its g option
Hi, With the g option, you just have to continue in the CALLER Dialplan, you have nothing to do, just continue your Dialplan i.e: exten= s,n,Dial(what you want) = and when the Called hangup you're goto the next line exten= s,n,Goto(where you want) or exten= s,n, 'DO WHAT YOU WANT: playback, background and so' After the CALLED party hangup (of course, not the caller), the CALLER continue in his dialplan.. Hope i'm not misunderstanding your question.. BUT if the two legs hangup, you have to use DEADAGI on the h extension.. - Original Message - From: voip crazy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 12, 2008 12:25 PM Subject: [asterisk-users] Dial command and its g option I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invitation to connect on LinkedIn
LinkedIn Josemar M#xfc;ller Lohn requested to add you as a connection on LinkedIn: -- Ricardo, I'd like to add you to my professional network on LinkedIn. -Josemar M#xfc;ller View invitation from Josemar M#xfc;ller Lohn http://www.linkedin.com/e/IUZTDdzrsg3rxGytdedLzTiomUEFOT3UdcnGbWCo8rrTM7G/blk/612574095_2/cBYRej0QdPkOcjoLqnpPbOYWrSlI/svi/ -- Learn how LinkedIn can power your career in 2008: static?key=promo_newyear_2008trk=300_8Tips_C -- (c) 2008, LinkedIn Corporation ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
On Thursday 12 June 2008 03:23:46 Mark Adams wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. And of course, the second part is choosing a secure password, one that contains mixed case, numbers, letters, and symbols. Don't be afraid to write down that secure password, as long as you keep it on your person (wallet is a good choice). 99% of the attackers who might otherwise compromise your machine will never come within 1000 miles of you. However, your wallet contains things that are far more valuable than your password (your identity documents, for example), so it is hoped that you will be able to keep that password away from people who would otherwise do you harm. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Philippe Sultan wrote:
Friends,
a new dialplan application is now available for testing :
http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
Sounds very cool.
See below for more comments:
The corresponding feature request is located here :
http://bugs.digium.com/view.php?id=12569
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id : ${CALLERID(all)});
Answer();
JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the
number you wish to call);
JabberReceive(${CALLERID(name)},NEWEXTEN);
How can you assume that the message you are waiting for is the right one
? Let's say that you have 10 channels each doing a JabberReceive at the
same time - how does the channel know how to get the right message, let
alone the right data ?
(2 channels may be waiting for a NewExten message, others for a
GetSomeDataFromSomeOtherPlace message )
Or am I missing something really obvious ?
Julian
JabberSend(asterisk-xmpp,$(CALLERID(name),(Calling ${NEWEXTEN} ...);
Dial(SIP/${NEWEXTEN);
Hangup();
}
}
In this example, when Asterisk receives a GoogleTalk voice call
request from a GoogleTalk buddy, it answers the call, and asks the
buddy to enter a number over an XMPP (Jabber) chat session. Then,
Asterisk dials the extension (accessible over SIP), which results in a
GoogleTalk to SIP call.
But this application is not restricted to GoogleTalk voice calls, and
it can be used within any call context. Code snippets are available in
the corresponding feature request under the bugtracker as well as in
doc/jabber.txt.
The codebase is Asterisk's SVN trunk, which is merged to the
jabberreceive branch on a regular basis. To install it, follow these
steps :
#svn co http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/
jabberreceive
#cd jabberreceive
#./configure
#make
#make install
Note for Linux users : the Gnome IM+ToIP client Empathy (starting from
version 0.23.1) is now compatible with Asterisk, which allows users to
place voice calls over a GoogleTalk channel from their Empathy client
to Asterisk.
Please give your feedback!
Thanks i advance,
Philippe
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Re: [asterisk-users] Asterisk Data Calls
On Thursday 12 June 2008 03:50:30 Tobias Wolf wrote: Tilghman Lesher schrieb: On Wednesday 11 June 2008 10:20:15 Brent Davidson wrote: There is not, although I don't see any reason why it couldn't be done. There is a ZapRAS application which performs much of this same function, although it only works on ISDN lines (where the line signal is already a digital stream of bits). But ZapRAS can only be used to dial-in with another ISDN Modem on my side, right. If i have a simple analouge modem there will be no data connection because of the different protocolls. Is this correct? Correct. ZapRAS is not a soft modem at all, but merely a bridge between an already digital channel and the network interface. If you had a PRI (ISDN) line and a dialup ISDN modem, it could conceivably be used for the function the OP needed. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] time on asterisk
I am also having this problem using includes based on time of day, however the restart did not help and when enabled it finds no context with extension 's'. This is for my incoming calls, see below...Any Ideas! [default] include =extensions include = after|18:00-7:29|mon-fri|*|* include = during|7:30-17:59|mon-fri|*|* ;include = after|*|sat|*|* ;include = after|*|sun|*|* ;always on;;; ;exten = s,1,Answer ;off ;exten = s,2,Background(/tmp/afterhours) ;on ;exten =s,2,dial(sip/202sip/203,20) ;exten=s,3,hangup() ;maingreeting [during] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten =s,3,Set(TIMEOUT(DIGIT)=3) exten = s,4,Background(/tmp/maingreeting) exten =s,5,waitexten(5) exten =s,6,dial(sip/201,20|t)) exten =s,7,hangup() exten = 1,1,Goto(dental,s,1) exten =1,2,hangup() exten = 2,1,dial(zap/g2/322) exten = 2,2,voicemail(u322)) exten =2,3,hangup() exten = 3,1,dial(sip/241,20) exten = 3,2,voicemail(u241) exten =3,3,hangup() exten = 4,1,Goto(medical,s,1) exten =4,2,hangup() exten = 6,1,dial(sip/232,20) exten = 6,2,voicemail(u232) exten =6,3,hangup() exten = 7,1,Goto(admin,s,1) exten =7,2,hangup() exten = 0,1,dial(sip/201,20) exten = 0,2,voicemail(u201) exten =0,3,hangup() ;;;Nightmode [after] exten = s,1,Answer exten =s,2,Set(TIMEOUT(DIGIT)=2) exten = s,3,Background(/tmp/afterhours) exten =s,4,waitexten(10) exten =s,5,hangup() exten =5,1,Dial(zap/g1/,20) exten =5,2,hangup() ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
On Thu, Jun 12, 2008 at 04:23:46AM -0400, Mark Adams wrote: My situation seems unique because I am not using a router even at this point. I was given a sheet of ip addresses and was told just to provision by devices with the given ip's and they would handle the rest. My devices are hooked directly to their switch in my location. This hasn't been an issue up until now because I only had analog (mediatrix and audiocodes 24 port gateways x 4) connected to the switch. Now I am going to a software based dialer (i.e. asterisk/ vicidial) and have run into these problems. This is one of the reasons why VoIP/Internet can be problematic: even if you have a firewall, you're required to expose your SIP or IAX ports to the net at large, whether through a firewall, or some sort of proxy -- which means you're at the mercy of people finding exploits in Asterisk that they can use to pwn your machine. Probably the only *really* good approach to this is the one we use here at Vici: don't let SIP and IAX out of the building. All of our PSTN connections are via traditional T-1 trunking to IXCs, and all of our agents are inside the building as well, on T-1/Zap/DAHDI channelbanks. If I ever do have to put people outside the building, I'll put them on secure VPNs, and the same if I have to trunk to commercial VoIP carriers. At the very least in this latter case, I'll IP lock the incoming connection, if I can't find a carrier that will do VoIP/VPN/Internet. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
On Thu, Jun 12, 2008 at 07:03:52AM -0400, Mark Adams wrote: I have a tellabs 8813 switch provided from time warner. No I currently do not have access to the switch. I am in the process of converting from analog based dialers using dialogic hardware TO asterisk/ vicidial systems I am strictly placing sip calls to my termination provider. I do not use the linux box for anything else. This fiber connection is dedicated to sip g729 calls entirely. Yes the fiber terminates directly to the switch. There are 6 analog to voip gateways (audiocodes and mediatrix) and 1 asterisk server. The gateways and 1 asterisk server are connected to the tellabs switch, security was never an issue because for the last 2 years we only connected analog to voip gateways to the open fiber connection. Ok, step one: Put a router/firewall behind that tellabs. It should be the only thing on your premises with a public routable address. Yes, that will make your SIP configuration a touch more complicated, but you've already seen the balancing cost... Go pick up a copy of Firewalls and Internet Security from O'Reilly, find a quiet corner and a pot of really *hot* tea, and sit and read it, assuming you can't pay someone else to do this for you. But don't put *nix application servers directly on the net unless you really know what you're doing. And yeah, you're gonna have to wipe that box, as Steve said. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote: On Thursday 12 June 2008 03:23:46 Mark Adams wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. Your phrasing, here, Tilghman, suggests that you mean that the administrative account should be renamed from root to madams, and I'm fairly sure you don't actually mean that. You actually mean create a regular user, and lock the machine down so that's the only thing that can be used to log into it, at which point, when and If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. ...you can use sudo to get it. And of course, the second part is choosing a secure password, one that contains mixed case, numbers, letters, and symbols. Don't be afraid to write down that secure password, as long as you keep it on your person (wallet is a good choice). 99% of the attackers who might otherwise compromise your machine will never come within 1000 miles of you. However, your wallet contains things that are far more valuable than your password (your identity documents, for example), so it is hoped that you will be able to keep that password away from people who would otherwise do you harm. Two memorable words separated by 2 or 3 digits, with at least one odd capital, is my usual protocol. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Command Option D Early Bridged
On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote: However, in my experience, the timing the call get bridged is not consistance, Do you happen to be calling out over an analog phone line? In the case of dialing out an analog line, we have no easy way of knowing when the far-end has answered the call, so the call is considered answered at the time the call is dialed. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Tilghman Lesher wrote: On Thursday 12 June 2008 03:23:46 Mark Adams wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. And of course, the second part is choosing a secure password, one that contains mixed case, numbers, letters, and symbols. Don't be afraid to write down that secure password, as long as you keep it on your person (wallet is a good choice). 99% of the attackers who might otherwise compromise your machine will never come within 1000 miles of you. However, your wallet contains things that are far more valuable than your password (your identity documents, for example), so it is hoped that you will be able to keep that password away from people who would otherwise do you harm. Most recent hacks that I have first or second hand knowledge of came from ssh issues. Most inexperienced admins will expose ssh without using the 'allowgroups' option in their sshd_config and will get hacked by someone logging in via ssh using a system account with no password. The second thing to do with ssh is to move it to another port to keep the script kiddies from pounding on it. If there is a weak or missing password, they will find it. An encrypted USB thumbdrive is also a good storage device for passwords. I use TrueCrypt and have the executable availble unencrypted on the thumbdrive so I could plug it into almost any machine and get to the encrypted data. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 phones, BRI and Analogue cards
On Thu, 2008-06-12 at 01:23 -0700, bilal ghayyad wrote: Where did u find a good IAX IP Phone? I've had good success with my Allnet IP-7960 phones. They have the ability in the firmware to either do SIP or IAX, and they even have a mode where you dial one prefix to send the call out using the SIP protocol, and another prefix to send the call out over the IAX protocol. They're not the best-looking phones in the world, but they seem to work quite well. More information (in German) at http://www.allnet.de/allsip/produkte/all7960.php -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Securing Asterisk and your network
On Thu, Jun 12, 2008 at 08:41:18AM -0500, Lyle Giese wrote: Most recent hacks that I have first or second hand knowledge of came from ssh issues. Most inexperienced admins will expose ssh without using the 'allowgroups' option in their sshd_config and will get hacked by someone logging in via ssh using a system account with no password. The second thing to do with ssh is to move it to another port to keep the script kiddies from pounding on it. If there is a weak or missing password, they will find it. This is true, and I'd forgotten to mention it. Update your machine regularly, and always take security updates, even if they cause breakage you have to chase down. Additionally, you should install a brute-force-attack blocker: http://www.la-samhna.de/library/brutessh.html I like the tcp_wrappers version, but whatever suits you. An encrypted USB thumbdrive is also a good storage device for passwords. I use TrueCrypt and have the executable availble unencrypted on the thumbdrive so I could plug it into almost any machine and get to the encrypted data. Though note that all currently extant hardware-secured thumbdrives are snake oil. I recommend Bruce Schneier's Password Safe (and not any of the other, similarly named programs) if you feel the need to store a lot of authentication credentials. Or get a BlackBerry and use theirs. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitoring QoS
Hi, While I haven't personally used any of their equipment yet, Brix is supposed to have good h/w and software for measuring a MOS score: http://www.brixnet.com/products/BrixCall.shtml http://www.voiptroubleshooter.com/basics/mosr.html -- James Hello Fellow Users, I am looking for a way - using certain software or other techniques - to monitor, measure, and improve the quality of service for Asterisk system. During the last while, it seems the quality has decreased and am trying to look for ways to get things going well again. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Thanks for all the help. I have been in this biz for several years using windows machines and analog dialers. I need to get on top of learning enhanced networking, linux systems and firewalls. Lots of goof information - Much appreciated! Mark Adams _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Thursday, June 12, 2008 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber Tilghman Lesher wrote: On Thursday 12 June 2008 03:23:46 Mark Adams wrote: I appreciate the responses thus far but I am looking to find out what type of security I should implement for the future. Being new to linux, not to mention asterisk I didn't realize that someone could brute force into the box and upload crap. With that in mind it seems that I would want to get a hardware firewall such as a hotbrick or a sonicwall firewall. One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. And of course, the second part is choosing a secure password, one that contains mixed case, numbers, letters, and symbols. Don't be afraid to write down that secure password, as long as you keep it on your person (wallet is a good choice). 99% of the attackers who might otherwise compromise your machine will never come within 1000 miles of you. However, your wallet contains things that are far more valuable than your password (your identity documents, for example), so it is hoped that you will be able to keep that password away from people who would otherwise do you harm. Most recent hacks that I have first or second hand knowledge of came from ssh issues. Most inexperienced admins will expose ssh without using the 'allowgroups' option in their sshd_config and will get hacked by someone logging in via ssh using a system account with no password. The second thing to do with ssh is to move it to another port to keep the script kiddies from pounding on it. If there is a weak or missing password, they will find it. An encrypted USB thumbdrive is also a good storage device for passwords. I use TrueCrypt and have the executable availble unencrypted on the thumbdrive so I could plug it into almost any machine and get to the encrypted data. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk Only as Voice Recording Solution.
HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: 1. Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. 2. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. 3. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on PRI even with H/W echo cancel
Joe, I'm unable to find an incident in your name in Digium Support's tracking systems. Can you email me off-list with a case or reference number, or give me contact information so one of our technicians can work with you to address your echo issue? A week-long wait is not typical of Digium Support. We strive to be as responsive as possible, and typically do much better than that. I'd like to understand what happened here, so we can make sure that we consistently provide quick and reliable support. Thanks, ~Kenny Shumard Digium Technical Support Manager - Original Message - From: Joe Pukepail [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, June 11, 2008 8:46:47 PM GMT -06:00 US/Canada Central Subject: [asterisk-users] Echo on PRI even with H/W echo cancel Hello, I have a PRI coming into a Digium TE122B with hardware echo cancel, but we are still experiencing echo on the first 10 seconds of a call. Is there anything that can be done about this? I have tried contacting digium support, but have not heard back from them (placed a support incident about a week ago). I see on digiums website that some of their card have a VPMOCT128 Octasic echo cancel, but the TE122B comes with digiums VPMADT032 echo canceler. Is the octastic echo cancel better? Should I look into a card with Octastic echo canceler? I see Sangoma has a single port with T1 with Octastic echo canceler or would have to move up to the dual span card to get the Octastic echo cancel on the digium card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice Recording Solution.
On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI after Hangup
Look for the DeadAGi command. Thanks l. On Thu, 12 Jun 2008 13:41:14 +0200, voip crazy [EMAIL PROTECTED] wrote: Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2 qualify problem - PONG ignored
Hello everybody I have a problem using the 'qualify' option with iax2: - snip -- Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 9ms SCall: 06558 DCall: 0 [213.235.242.217:4569] Jun 12 16:11:14 WARNING[22657] chan_iax2.c: Received mini frame before first full voice frame Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: VNAK Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 00011ms SCall: 06558 DCall: 00303 [213.235.242.217:4569] Jun 12 16:11:14 VERBOSE[22657] logger.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 9ms SCall: 00137 DCall: 06558 [213.235.242.217:4569] Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 0ms SCall: 06558 DCall: 00137 [213.235.242.217:4569] Jun 12 16:11:14 VERBOSE[22657] logger.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 9ms SCall: 00137 DCall: 06558 [213.235.242.217:4569] Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 0ms SCall: 06558 DCall: 00137 [213.235.242.217:4569] Jun 12 16:11:14 WARNING[22657] chan_iax2.c: Received mini frame before first full voice frame Jun 12 16:11:14 VERBOSE[22657] logger.c: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: VNAK Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 00030ms SCall: 06558 DCall: 00303 [213.235.242.217:4569] Jun 12 16:11:14 VERBOSE[22657] logger.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Jun 12 16:11:14 VERBOSE[22657] logger.c:Timestamp: 0ms SCall: 00303 DCall: 06558 [213.235.242.217:4569] Jun 12 16:11:18 NOTICE[22657] chan_iax2.c: Peer 'iax-sil' is now UNREACHABLE! Time: 5 snip Obviously the PONG response(s) are ignored (even answered with INVAL) and the peer goes unreachable! Seems like a bug to me (could it be related to the out-of-order miniframe/VNAK?) I'm using Asterisk v1.2.13 (Debian stable). Is there a solution/patch to fix this? -- Stephan Weinberger [EMAIL PROTECTED] pgpjHfuFJSt5a.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI after Hangup
You have to run DeadAGI, in h . Regards Andrea Cristofanini voip crazy ha scritto: Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Jay R. Ashworth wrote: On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote: One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. Your phrasing, here, Tilghman, suggests that you mean that the administrative account should be renamed from root to madams, and I'm fairly sure you don't actually mean that. You actually mean create a regular user, and lock the machine down so that's the only thing that can be used to log into it, at which point, when and If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. ...you can use sudo to get it. Never, ever, ever, expose sshd to the public internet without firewalling. Only let trusted IPs reach sshd. The risk of brute force success, however small, is still far too great. Again, do not expose sshd to the general public. And for that matter... it's generally unwise to expose any service to the general public when the general public has no business using that service. A little bit of time learning some iptables basics will go a long way here. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Hi Julian,
[...]
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id : ${CALLERID(all)});
Answer();
JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the
number you wish to call);
JabberReceive(${CALLERID(name)},NEWEXTEN);
How can you assume that the message you are waiting for is the right one
? Let's say that you have 10 channels each doing a JabberReceive at the
same time - how does the channel know how to get the right message, let
alone the right data ?
(2 channels may be waiting for a NewExten message, others for a
GetSomeDataFromSomeOtherPlace message )
Well, in the example, as long as you have 10 simultaneous GoogleTalk
calls from 10 different buddies, that won't be a problem. The first
argument of JabberReceive is used by the channel to identify the
Jabber ID it expects to read data from. Therefore, a message coming
from a specified buddy (identified by his JID) will be passed by
res_jabber to the channel that is waiting for data from this buddy.
In the case when several channels are waiting for data from the same
JID, res_jabber passes the message to every channel that matches.
Although this is less likely to happen, I tried to address this issue
by using the thread tag to track chat conversations
(http://www.xmpp.org/extensions/xep-0201.html). Unfortunately, very
few XMPP clients implement this conversation tracking mechanism (and
GoogleTalk does not).
Philippe
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Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber
Yes it all makes sense, I left it all open so sip traffic could pass. My experience has only been with analog gateways which well no one would wasn't to break into or do any of these things too. Thanks for the sonicwall tip, that was what I was about to buy. Mark Adams -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Thursday, June 12, 2008 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] aSTERISK / Vicidial systems over 4MB fiber Jay R. Ashworth wrote: On Thu, Jun 12, 2008 at 08:02:24AM -0500, Tilghman Lesher wrote: One of the most frequent security issues comes not in the form of a software flaw, but simply in people choosing easy-to-guess passwords on the root account. There are two suggestions I have to reduce the risk of this brute force. First, choose a username that is uncommon. In your case, do not use 'root', 'admin', or even 'mark'. 'madams' might be a good choice. Once you figure out that username, configure sshd with the AllowUsers directive to ONLY allow logins from that user. Your phrasing, here, Tilghman, suggests that you mean that the administrative account should be renamed from root to madams, and I'm fairly sure you don't actually mean that. You actually mean create a regular user, and lock the machine down so that's the only thing that can be used to log into it, at which point, when and If you need root access, install sudo. If an attacker cannot figure out what your username is, then it doesn't matter even if they guess your password, because they aren't getting in. ...you can use sudo to get it. Never, ever, ever, expose sshd to the public internet without firewalling. Only let trusted IPs reach sshd. The risk of brute force success, however small, is still far too great. Again, do not expose sshd to the general public. And for that matter... it's generally unwise to expose any service to the general public when the general public has no business using that service. A little bit of time learning some iptables basics will go a long way here. Thanks, Lee. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and XMPP (Jabber) : testing new application JabberReceive
Hi Philippe,
thanks for the replies. It all seems sensible.
Now, for a request ;)
How difficult would it be to have a JabberReceive Event *initiate* a
channel ?
This could be done by specifying a [EMAIL PROTECTED] in
jabber.conf
So, when a message is received by asterisk, a call is initiated in the
extension and context defined, and have the jabber message details in
variables ${JABBERFROM}, ${JABBERMESSAGE} etc etc.
It would also be really useful to be able to initiate AMI commands via
jabber ;)
Julian
Philippe Sultan wrote:
Hi Julian,
[...]
What can you do with it? Well, a direct usage of this application is
to make an easy to use GoogleTalk voice gateway out of Asterisk. Here
is an example (assuming the asterisk-xmpp account is configured) :
context gtalk-in {
s = {
NoOp(Caller id : ${CALLERID(all)});
Answer();
JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the
number you wish to call);
JabberReceive(${CALLERID(name)},NEWEXTEN);
How can you assume that the message you are waiting for is the right one
? Let's say that you have 10 channels each doing a JabberReceive at the
same time - how does the channel know how to get the right message, let
alone the right data ?
(2 channels may be waiting for a NewExten message, others for a
GetSomeDataFromSomeOtherPlace message )
Well, in the example, as long as you have 10 simultaneous GoogleTalk
calls from 10 different buddies, that won't be a problem. The first
argument of JabberReceive is used by the channel to identify the
Jabber ID it expects to read data from. Therefore, a message coming
from a specified buddy (identified by his JID) will be passed by
res_jabber to the channel that is waiting for data from this buddy.
In the case when several channels are waiting for data from the same
JID, res_jabber passes the message to every channel that matches.
Although this is less likely to happen, I tried to address this issue
by using the thread tag to track chat conversations
(http://www.xmpp.org/extensions/xep-0201.html). Unfortunately, very
few XMPP clients implement this conversation tracking mechanism (and
GoogleTalk does not).
Philippe
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Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
You will need exactly two times the number of ports that your legacy system has. Asterisk takes the call on _.,1,DAHDI, starts monitor and dials out the second DAHDI port to your legacy system. It is about ten lines in extensions.conf. Thanks, Steve T On Thu, Jun 12, 2008 at 12:01 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to turn on the H323 logging on Asterisk
Hi Sema,
In article [EMAIL PROTECTED],
Sema Arca [EMAIL PROTECTED] wrote:
Thanks a lot for the tips. I have turned on the logging and saw them in the
console. However, this applies only for startup, when I try to register a
user, which I cannot succeed, there is no logging done.
Check your logger.conf file and make sure that verbose is going either to
messages, or full or both. Then you will be able to extract relevant debug
output for any future posting.
Do you think you can give me an idea why my user cannot register? All I want
to do is, send the register to a GK which has the IWF functionality.
I haven't enough information to answer that, and I am not very familiar
with gatekeepers anyway.
Attached I am sending my extensions.conf and ooh323.conf files.
A few things I noticed:
In your extensions.conf, you have this:
[h323]
exten = _1NXXNXX,1,Dial,H323/${EXTEN}
When using chan_ooh323, you need to use OOH323 instead of H323, and you
also need to name the peer you want to dial out through:
exten = _1NXXNXX,1,Dial(OOH323/[EMAIL PROTECTED])
Or possibly Sema instead of mypeer1, depending on what you need.
In your ooh323.conf, you should disallow all codecs first, and then just
allow the ones you want. For initial testing, just stick with ulaw or alaw:
disallow=all
allow=ulaw
allow=alaw
Then remove the allow=all from your peer, user and friend sections.
There may well be other issues, but fix these first. Then if you still
get problems, include the verbose output from the log file.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Fwd: Complimentary Subscription to VoIP Industry Publication
So is this Digium taking over Pulver's' void? Same color scheme and fonts. Thanks, Steve -- Forwarded message -- From: Digium [EMAIL PROTECTED] Date: Thu, Jun 12, 2008 at 10:34 AM Subject: Complimentary Subscription to VoIP Industry Publication To: [EMAIL PROTECTED] If you are having trouble reading this email, read the online version. Complimentary Subscription to VoIP Industry Publication If you are in the business of IP communications, then you will be interested in keeping track of trends and developments in the sector. Due to your association with Digium(R), we are pleased to offer you a complimentary charter subscription to FierceVoIP. FierceVoIP is a twice-weekly e-mail briefing on the entire IP communications industry including VoIP business, IP technology, VoIP security, unified communications, and more. Save time and keep up to speed on: IP service provider and enterprise vendor news Industry trends, stats, and developments Innovative technologies, standards, and regulations New products, deals, deployments and investments Movements of people and players in the industry Sign up for your complimentary subscription to FierceVoIP: What's Next in IP Communications Join over 42,500 industry insiders who depend on FierceVoIP for their twice-weekly VoIP industry e-mail briefing. Make sure you are not missing out. We look forward to you taking advantage of this special complimentary offer and welcome you to the FierceVoIP family of industry leaders Recent coverage in FierceVoIP: Nortel's Return From Doom Buy Your IP PBX at Costco ShoreTel shouts more UC, new switches Helpful Hints for VoIP SME Mobile SIP to Surpass IMS? RESERVE MY COMPLIMENTARY SUBSCRIPTION NOW Best, FierceVoIP Circulation Team About this email: If you do not want to receive messages from FierceVoIP, email [EMAIL PROTECTED] with UNSUBSCRIBE in the subject line. Copyright (c) Digium, Inc. - The Asterisk Company, 445 Jan Davis Drive NW, Huntsville, AL 35806 Visit our website: Digium.com | Unsubscribe | Update Subscription ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI after Hangup
Thanks for your answers, DeadAGI was the solution. Thanks again. Voipcrazy 2008/6/12 Andrea Cristofanini [EMAIL PROTECTED]: You have to run DeadAGI, in h . Regards Andrea Cristofanini voip crazy ha scritto: Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invitation to connect on LinkedIn
Fail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] custom functions is voicemail
Hi, I want to add some custom functions in voicemail. For example user can switch SMS on/off or the voicemail global on/off. Whats best way to do this? modify app_voicemail.c or or do everything in dialplan? or any other solutions (Asterisk 1.2.X please) best regards Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] On Hold Context?
Hi list, I have some on hold activities that I would like to implement and I'm just wondering if there is a way to do it. Here's what I'm thinking: While a caller is on hold, they could have the option to do things like retrieve weather, news, play blackjack( ala tellme), etc. These are all things I could do in the conventional method of sending someone to a context, but I have no way to pull them out when their call is ready to be taken. As far as news, weather and any recorded info, I could theoretically do this by simply retreving the info and making an on hold class that used that audio, but is there anyway that would allow interaction (that doesn't involve tossing the caller into an empty conference room)? Thanks! -Kris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21 Released
The Asterisk.org development team has released Asterisk version 1.4.21. This release is a regular bug fix release for the 1.4 series of Asterisk. For a full list of changes, see the ChangeLog included in the release. * http://svn.digium.com/view/asterisk/tags/1.4.20/ChangeLog?view=markup Asterisk 1.4.21 is available for immediate download from the Digium downloads site. * http://downloads.digium.com/pub/telephony/asterisk/ Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] New faxing protocol. Good/Bad ?
Dovid Bender wrote: Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or VOIP fax machine would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and get a conformation on it but maybe this would work a bit better. Send the entire image over and then get a response when it is done. This way if there is issues along the way the packets can be re-sent with out any issue. Dovid (Cross posted to Biz for those that aren't on the users list - I want their onion too ;) ). Hi For those interested in above, PBXware had it implemented 2 years ago. Regards, Senad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Unified communication features
Hi, I need to know if the following features are available on asterisk and their quality -SMS -Call control, budgeting and monitoring -Video conferencing -support for 500 extensions -fax -audio and video conferencing and 1. Call accounting showing calls made 2. Call budgeting which bills the calls 3. Web access for all users 4. Centralized management and administration 5. Call barring when budget is exhausted 6. Budget utilization alerts to e-mail 7. Reports a) Per extension b) Per trunk c) Per unit (business area) d) Percentage utilization of the total budget Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems getting dialed information on asterisk
Hello I have asterisk with a sip connection with an external provider, getting the calls from a toll free number. Asterisk collects information from the callers, using also an script on windows with the dial plan. Debuging the script, I noticed that randomly asterisk losses some of the characters I typed on the phone. ?Is there anyway to avoid this problem? Thanks in advance for any advice Greetings Enediel _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phone selective variable setting?
Hi Forum, nbsp; While integrating a Nokia E61i cell phone into my Asterisk installation, I have encountered an issue that I have pin-pointed to the phone's SIP protocol. nbsp; Upon the arrival of an incoming call, the dialplan set the variable CALLERID(name) to the caller's name, then it dials a bunch of telephones (Nokia is one amongst them) with one unique DIAL command. nbsp; While the three phones behave correctly by ringing, the E61i replies with a nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp; [Got SIP response 400 Bad Request back from..] error message, then Asterisk notifies that SIP/5123-cb06 is circuit-busy. nbsp; I've raised the issue with Nokia, but in the meantime I'm looking for a workaround in the dialplan that will allow me to dial all phones, setting the CALLERID(name) variable for those phones that behave themselves while not setting it for Nokia's. nbsp; Is there a way to do that? How? Other options? nbsp; Thanks for your help, nbsp; nbsp; Costa nbsp; ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? I believe that ForkCDR is the solution: ForkCDR Synopsis Forks the Call Data Record ForkCDR() Causes the Call Data Record to fork an additional cdr record starting from the time of the fork call. Description Fork The CDR into 2 separate entities. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
yeah something like that. is it possible to set asterisk to make 10 calls per second? On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote: I know you can limit the total calls in any given time, for example, you say I would like to have 10 SIP calls established as maximum. On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote: Is there a way to limit or set the calls per second on SIP? -- Regards, Mark Quitoriano Blog | http://mark.quitoriano.org VicidialNOW! | http://www.vicidialnow.com APUG! | http://asterisk.org.ph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mark Quitoriano Blog | http://mark.quitoriano.org VicidialNOW! | http://www.vicidialnow.com APUG! | http://asterisk.org.ph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial command and its g option
snip BUT if the two legs hangup, you have to use DEADAGI on the h extension.. Quick note, he doesn't necessarily have to use DeadAGI unless it's an AGI being called. He just has to make sure he defines the h extension in that context and set up the same executions as the post-dial executions (if AGI is used, yes you need DeadAGI). Just thought I'd clear it up -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
On Thu, Jun 12, 2008 at 9:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
Combine GROUP/GROUP_COUNT with category of ${EPOCH}
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group
Calls will still be received by asterisk, however you will be able to
kick them off without proceeding with following dialplan logic.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] asterisk calls per second
Well, as I said, you can tell Asterisk to accept until 10 SIP calls,
for example, at ANY TIME (I don't understand why per second, I mean,
if the 10 calls are established in the same second, they are acepted,
and so they are if they are established in the same milisecond, while
the max concurrent calls is below the limit of 10).
You can do something like this in your dialplan (assuming extensions like _3XX)
exten=_3XX,1,Set(GROUP()=sip-calls)
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(sip-calls)})
exten=_3XX,3,GotoIf($[${GROUPCOUNT} ${MAX_CALLS}]?120)
exten=_3XX,4,Dial(SIP/${EXTEN})
exten=_3XX,5,Playback(unavailable)
exten=_3XX.,6,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
where ${MAX_CALLS} is a variable defined by you that is the limit of
calls to be accepted
On Thu, Jun 12, 2008 at 12:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
Atis Lezdins wrote: On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis Ah thanks Atis! I hadn't played with it before since the documentation gave info that lead me to believe it wouldn't work for me :) Very helpful information :) -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Securing Asterisk and your network
On Thu, Jun 12, 2008 at 09:53:53AM -0400, Jay R. Ashworth wrote: Additionally, you should install a brute-force-attack blocker: http://www.la-samhna.de/library/brutessh.html This is effectively another service listening. It is also a method for an attacker to lock you out of the system. See, for instance, http://www.ossec.net/en/attacking-loganalysis.html . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Really destroying SIP dialog
I am trying to work in the console, figuring why it exits, but about 75% is always taken up with Really destroying SIP dialog '' Method: OPTIONS Can anyone point me where I can stop this without turning down the debugging/verbose on the entire console. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd Polycom Reboot Issue
Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every time it gets a call. As soon as the call hits the phone, a small blip is heard from the speaker, then the reboot is initiated. There is nothing shown in the asterisk logs to indicate the problem. Likewise, the logs sent by the phone via tftp are equally as useless. We've formatted the phone's filesystem causing it to get a fresh reflash of the firmware from tftp upon bootup. Same problem. Has anyone experienced an issue such as this? How should I proceed to diagnose and repair the problem? Thank you!! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi question
Hello, I'm wondering about following DUNDI setup Suppose we have 2 Asterisks: astA and astB with DUNDI peering active between them and 2 SIP endpoints: sipA registered with astA and sipB regsitered on astB All this is on the same LAN now sipA call an number which corresponds to [EMAIL PROTECTED] , so astA lookups thru DUNDI at astB and forwards the call there. My question is how this fowarding is done ? Using SIP RE-INVITE, or REFER, or using SIP 301 responce with Contact pointing at [EMAIL PROTECTED] And does the final RTP stream traverse both Asterisks or only one of them or None of them? Thanks Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help-ASTERISK-MFCR2
Hi guys, here I attach a call log made by Mariano (full log enabled in logger.conf, set debug 100, set verbose 100 on asterisk console)... 1) first call, is one from pstn (EWSD) to our asterisk box. http://rafb.net/p/HCoodb28.html 2) 2nd call, is one from a sip user registered at asterisk, to a the EWSD switch http://rafb.net/p/l6PKMI14.html with those calls are attached too the unicall.conf, zaptel.conf and an snippet of the extensions.conf If you need anything more to can help us please tell me.. thanks.. On Tue, Jun 10, 2008 at 9:29 AM, Mariano Borgognone [EMAIL PROTECTED] wrote: Alvaro, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log notice,warning,error,debug,verbose). The log console output is: Here is the LOGS when I try do make calls, the call will not go to Asterisk Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [2/ 2/Seize ack /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/1000/Clear fwd /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release guard expired Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32769 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Thanks ... Regards, Mariano - Original Message - From: Alvaro Parres To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 09, 2008 7:04 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 Mariano: Could you send us please the log files, and the console output... so we can help you. On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone [EMAIL PROTECTED] wrote: Moises, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log (notice,warning,error,debug,verbose). Has anyone experienced with a Siemens EWSD switch? Anyone knows about to change R2 timers at unicall.conf ? Please any comment is welcome, thank you.. Mariano.- - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, June 07, 2008 1:27 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone [EMAIL PROTECTED] wrote: Dears, I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive call for PSTN, i don´t understand why. please i need your help # MFC/R2 normalmente no usa CRC4 span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 loadzone=us defaultzone=us [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes
[asterisk-users] Astricon question: four or five tracks?
We're busily churning away at creating the Astricon (http://www.astricon.net/) talk track this year, and it's been delayed by a problem that we've never had in years past: too many high-quality talk submissions. Not a bad problem to have, but still a problem. We have four tracks on the schedule: 1) Business Track - this relates to things like creating business models around Asterisk, technologies that embed aspects of Asterisk into their platforms, discussions of open source in the marketplace, and new technologies that can be added to Asterisk for specific application delivery reasons, among other topics. 2) Technology Track - Intro/Intermediate - Topics here range from basic introductions to Asterisk as far as feature sets and capabilities, and even into the moderately challenging topics of introductions to embedded systems and case studies. 3) Technology Track - Advanced - This includes more advanced implementation studies, protocol topics, new Asterisk features (LUA, for example), and inner workings of various Asterisk and third-party components. 4) Technology Track - Call Center/Large Scale - More case studies here but focused on large-scale systems. Carrier issues such as call recording, conferencing, clustering, and call center topics. We have had an overwhelming number of top-notch technical submissions for talks this year, which has been GREAT. Last year, we heard that there was a desire for even more technical tracks, so this year will fulfill that need. But we're stuck - we have way more topics than we have slots in the 4-track schedule, and so we've hit an impasse. We've had to start looking at cutting some really interesting topics because we simply don't have the space in the schedule. This is a terrible position, and so we're looking for what we can do to fix the problem. The obvious choice is Well, why don't you add a fifth track? So that is why I'm putting this message out. It's possible for us to add a fifth advanced technical track, but that would mean that there would be at any one time FIVE talks happening, four of which would be technical, and three of which would be classified as advanced. It will certainly be the case that there are overlapping areas of interest. Even with a fifth track, we are STILL going to have to turn down a few of the requests in the queue because of lack of slots, and at this point extending the conference another day is a very difficult option due to the hotel scheduling which is done far in advance. We also had some feedback from years past that a two-day conference seemed to suit everyone's schedules better, so this may be some unintended consequences from the compression. Our question to the community is: Is it too much to have 5 talk tracks at Astricon? Our initial instinct is Go ahead and do it but this does sound like a question that should be posed to the people who will attend. Your opinion would be valued if you could take the time to reply, but please try to summarize at the top of any replies with a Yes or No (even if you have more things to say) so I can keep a bit closer eye on the reply volumes. Feel free to reply on or off list. JT -- -- John Todd [EMAIL PROTECTED] Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on SLOW solid state disk
On Thu, Jun 12, 2008 at 3:23 AM, OCG Technical Support [EMAIL PROTECTED] wrote: I'm looking at building up a standard asterisk system fanless/no moving parts. I found a cheap solid state disk (Transcend TS32GSSD25S-M), but it is SLOW...25mb/sec read 8mb/sec write. I'm developing an asterisk based PBX on a TRASCEND DOM (ssd). At boot all seems slow, but since Linux kernels use higly buffered read/write operations, i will not have problems at all. But, consider to: 1. Use many many RAM. Astersik use few RAM but huge RAM is needed for disk buffers. 2. Screw the system and avoid all unnecessary writes on disk: logs, db, recordings, etc. 3. Do NOT use a journaled filesystem: i use ext2 3. Avoid swap (see 1) Bye. PicoStreamer - the real WEB live streaming software vinz486.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Browser based VoIP client?
Hilary Miller hilfmil at gmail.com writes: Something that I can put on our internal company website to replace our hardware IP phones. I see many web 2.0 startups offering browser based clients for their own service, but I can't seem to find anything that I can use with my own PBX. Do I suck at searching google or has the future not arrived yet? Thanks for reading! Take a look at http://www.mbdsys.com/opensource/veronix Vadim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon question: four or five tracks?
Hello, I would recommend that if you do add another tech track that you spend a great deal of effort trying to make sure that sessions that would appeal to similar audiances are not done at the same time. This has happened a few times in past Astricons and it's always a tough choice for attendees that are interested in both talks to choose between them. To this end, I might suggest even video-recording the presentations to be replayed at night during the conference(or possibly on the web) so attendees can see what they missed if they were unable to sit in on a presentation. One other suggestion I might make is that after 6PM I think there might be a benefit from loosly structured BOF or discussion sessions. There is only so much Red Bull and Alcohol you can drink in the code zone. I quickly organized two after-hours discussion sessions during last year's Astricon and actually had a few dozen people involved in each one, it would be great if this could be done on a larger scale and officially organized. Thanks, MATT--- On 6/12/08, John Todd [EMAIL PROTECTED] wrote: We're busily churning away at creating the Astricon (http://www.astricon.net/) talk track this year, and it's been delayed by a problem that we've never had in years past: too many high-quality talk submissions. Not a bad problem to have, but still a problem. We have four tracks on the schedule: 1) Business Track - this relates to things like creating business models around Asterisk, technologies that embed aspects of Asterisk into their platforms, discussions of open source in the marketplace, and new technologies that can be added to Asterisk for specific application delivery reasons, among other topics. 2) Technology Track - Intro/Intermediate - Topics here range from basic introductions to Asterisk as far as feature sets and capabilities, and even into the moderately challenging topics of introductions to embedded systems and case studies. 3) Technology Track - Advanced - This includes more advanced implementation studies, protocol topics, new Asterisk features (LUA, for example), and inner workings of various Asterisk and third-party components. 4) Technology Track - Call Center/Large Scale - More case studies here but focused on large-scale systems. Carrier issues such as call recording, conferencing, clustering, and call center topics. We have had an overwhelming number of top-notch technical submissions for talks this year, which has been GREAT. Last year, we heard that there was a desire for even more technical tracks, so this year will fulfill that need. But we're stuck - we have way more topics than we have slots in the 4-track schedule, and so we've hit an impasse. We've had to start looking at cutting some really interesting topics because we simply don't have the space in the schedule. This is a terrible position, and so we're looking for what we can do to fix the problem. The obvious choice is Well, why don't you add a fifth track? So that is why I'm putting this message out. It's possible for us to add a fifth advanced technical track, but that would mean that there would be at any one time FIVE talks happening, four of which would be technical, and three of which would be classified as advanced. It will certainly be the case that there are overlapping areas of interest. Even with a fifth track, we are STILL going to have to turn down a few of the requests in the queue because of lack of slots, and at this point extending the conference another day is a very difficult option due to the hotel scheduling which is done far in advance. We also had some feedback from years past that a two-day conference seemed to suit everyone's schedules better, so this may be some unintended consequences from the compression. Our question to the community is: Is it too much to have 5 talk tracks at Astricon? Our initial instinct is Go ahead and do it but this does sound like a question that should be posed to the people who will attend. Your opinion would be valued if you could take the time to reply, but please try to summarize at the top of any replies with a Yes or No (even if you have more things to say) so I can keep a bit closer eye on the reply volumes. Feel free to reply on or off list. JT -- -- John Todd [EMAIL PROTECTED] Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] funny search engine terms
Lol - I was checking the analytics for my www.collins.net.pr/blog site this afternoon and saw a funny search engine referral term - check out the search words in the output below. Funny how my site comes up first for that particular combination of words. I hope he/she found what they were looking for. BTW in case this person subscribes to the list as well - hopefully a few people will reply here with answers to your question.any thoughts? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.cognation.net/ Domain Name (Unknown) IP Address82.194.62.XXX (Batelco) ISP Batelco Continent :Asia Country :Bahrain (Facts) State/Region :Al Manamah City :Manama Language English (U.S.) Operating System Microsoft WinXP Browser Internet Explorer 7.0 Monitor Resolution:1280 x 960 Color Depth :32 bits Time of Visit Jun 12 2008 6:22:01 pm Time Zone UTC+2:00 Visitor's TimeJun 13 2008 1:22:01 am Referring URL http://www.google.com/search?hl=enq=personally%20how%20can%20i%20benefi t%20from%20asterisk Search Engine google.com Search Words personally how can i benefit from asterisk Visit Entry Page http://deancollinsblog.blogspot.com/2008/05/open-letter-to-asterisk-comm unity.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon question: four or five tracks?
On Thu, Jun 12, 2008 at 7:57 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, snip To this end, I might suggest even video-recording the presentations to be replayed at night during the conference(or possibly on the web) so attendees can see what they missed if they were unable to sit in on a presentation. snip I was very surprised that presentations were not video taped or at the least recorded at the last Astricon. I agree with Matt, choosing between even different topics or tracks can be difficult let alone similar topics. Recording almost seems like a no brainer, this is Asterisk after all. All attendees could probably cough up a little extra for the DVD if need be. It could also be sold I guess, but I would rather see the videos on YouTube or AsteriskTV or whatever free outlet. Thanks, Steve Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
hi sir, i forgot to mention it was originally at Asia/Singapore, when i noticed that asterisk has a wrong time, that's why i tried GMT instead. regards, Ron --- On Thu, 6/12/08, Stelios Koroneos lt;[EMAIL PROTECTED]gt; wrote: From: Stelios Koroneos lt;[EMAIL PROTECTED]gt; Subject: RE: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 9:33 AM GMT timezone does not have daylight savings, so probably this is why you have the wrong time Select a timezone for a city and usually the correct daylight parameters are used nbsp; Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com nbsp; From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Thursday, June 12, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] time on asterisk hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 lt;[EMAIL PROTECTED]gt; wrote: From: mkn0014 lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: gt; Hi Sir, gt; gt; I tried restarting asterisk, but still it has the wrong time. gt; gt; I tried restarting the system, then start asterisk it still uses the gt; wrong time. gt; gt; I also tried recompiling asterisk, checked i have the correct time on gt; the system, then restart the system then start asterisk but still i gt; get the wrong time. gt; gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008 gt; gt; on asterisk i use EPOCH to look at the time, gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) gt; gt; i would really appreciate any help. TIA gt; gt; ron gt; gt; --- On *Thu, 6/12/08, Tilghman Lesher gt; /lt;[EMAIL PROTECTED]gt;/* wrote: gt; gt; From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; gt; Subject: Re: [asterisk-users] time on asterisk gt; To: Asterisk Users Mailing List - Non-Commercial Discussion gt; lt;asterisk-users@lists.digium.comgt; gt; Date: Thursday, June 12, 2008, 1:42 AM gt; gt; On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; gt; between the asterisk time and the system time. could it be because i gt; gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; gt; of timezone on the system? How can I fix this prob? gt; gt; Yes, that's probably the reason. The system timezone is cached once at gt; startup, for performance reasons. The only way to get it to pick up the new gt; timezone is a restart. gt; gt; -- gt; Tilghman gt; gt; Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. You should actually be using Asia/Singapure rather than guess. i'm not using ntp, coz when i do i also don't get the correct time. That's because you have an incorrect timezone set. I am also using gotoiftime in my IVR but I don't have any problems. 1) Install the distro and specify the timezone 2) Set the correct time in linux 3) Install ntp 4) Sync the time by ntpdate ntp will always just sync using GMT time but the timezone specified in the distro will provide the time difference and daylight savings. That is it! Also, can someone clarify if Asterisk really uses a different time than the system time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
Hi Steve, thanks for your response... I will try it this saturday and I'll let you know... Best regards On Wed, Jun 11, 2008 at 7:11 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote: Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 Is your Asterisk box dual homed? Firewalled? Any output from the CLI with verbose turned on, that might help? Turn on SIP debugging as well. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nacho Linux Counter #156439 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT ??
Any suggestions ?
Available options for the two settings similiar to the one identified are as
follows:
admin set send-dnis-type-of-number?
send-dnis-type-of-number:
Type of Number to be sent in called party IE in the setup message to
pstn. For ISDN signaling. To be used on egress gateway for VoIP calls.
Enumerated field, values:
unknown:
international:
national:
network-spec:
subscriber:
abbreviated:
transparent: Setting this, we can pass TON transparently as received from
upper layers or in case of VoIP, as received from Near End gateway.
admin set send-dnis-numbering-plan?
send-dnis-numbering-plan:
Numbering Plan to be sent in called party IE in the setup message to
pstn. For ISDN signaling. To be used on egress gateway for VoIP calls.
Enumerated field, values:
unknown:
isdn-telephony:
data:
telex:
national:
private:
transparent: Setting this, we can pass NP transparently as received from
upper layers or in case of VoIP, as received from Near End gateway.
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Leon Sun [EMAIL
PROTECTED]
Sent: Monday, June 09, 2008 1:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
It should work.
Leon Sun
Times Telecom
Tel: 604-279-8787 ext 1586
Fax: 604-278-2793
Mobile: 604-780-2668
Click this button now and leave your phone number. Talk to me for free.
powered by www.clicksaya.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
We are providing voip services, these 911 calls are going out from our
subscribers to the lec to be delivered to the 911 PSAP.. Would this apply
in that scenario ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leon Sun
Sent: Sunday, June 08, 2008 3:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Joe,
I am not sure if your 911 call is incoming or outgoing on PRIs.
#assume you have a T1 in {1 1 1}
Read t1 { 1 1 1}
Set line send-dnis-type-of-number ?
You will see options. Some 911 providers support media-before-connect. Plz
make sure your all of TNT support 183.
Hope it can help you
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Sunday, June 08, 2008 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Alex.. would you point us in the right direction, or perhaps consider
sending a sample max tnt config reflecting how this is done? Thank you..
-Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, June 06, 2008 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
I believe the ISDN call plan can be configured as part of the trunk
group / route.
Joe Carroll wrote:
We talked with the LEC and discovered that 911 has to be sent as Unknown
instead of National... Anyone know how we might tell the TNT to do this?
Apparently, according to the carrier, all Special Access Numbers, 411, 611,
911, etc require this special code ???
PRI DEBUG FOLLOWS:
--nt SETUP CRV=14997 (Orig) Prot=Q931 12:51:47.260 06-06-08
Bearer_Cap 80 90 A2 (Speech,Rate=64K)
Channel_Id A1 83 83 (Pref,Intf=0,Chan=3)
Calling_Num (National,Restricted,Failed) 229317
Called_Num (National) 911
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Carroll
Sent: Thursday, June 05, 2008 6:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 911 via MAX TNT ??
Yes, we are using the max tnt to aggregate several PRIs both inbound and
outbound from multiple carriers. This PRI is a normal two way circuit that
a carrier would deliver to an end user...
From: [EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
[EMAIL PROTECTED]
Sent: Thursday, June 05, 2008 9:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 911 via MAX TNT ??
On Wed, Jun 04, 2008 at 08:07:18PM -0400, Andrew Kohlsmith (lists) wrote:
On June 4, 2008 06:20:57 pm Joe Carroll wrote:
Interestingly enough, on the syslog messages from the TNT we are seeing
Called = 911, Q850 Cause = 28, SIP Response = 484
That really looks like the switch that the TNT is talking to is rejecting
the
number, not the TNT...
Remember: 9-1-1 is a *dialling pattern*, not a *directory number*;
it's entirely possible that trunks wouldn't accept it directly.
This *is* a *LEC* trunk,
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
