Re: [asterisk-users] time on asterisk
Hi, I don't know what i'm doing wrong but i already reinstalled the system. still using ubuntu 64-bit. made sure i had the correct local date time. then did all this: ntpdate pool.ntp.org tzselect , i chose Asia/SIngapore /etc/timezone is Asia/Singapore i added TZ='Asia/Singapore'; export TZ to /etc/profile. date shows the correct date, i rebooted, date still shows the correct date. then installed zaptel, ./configure, make menuselect, make make install make config then libpri make make install then asterisk, ./configure, make menuselect, make, make install, make samples then asterisk-addons, ./configure make menuselect, make make install make samples then run asterisk, then connect via asterisk -r [Jun 13 00:25:58] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found [Jun 13 00:26:02] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found log shows June 13 00:25 my system date shows /home/ronald# date Fri Jun 13 15:33:03 SGT 2008 i installed everything as root via sudo su, how come i still dont get the correct time? really need help on this one. thank you regards nhadie --- On Fri, 6/13/08, Lee, John (Sydney) [EMAIL PROTECTED] wrote: From: Lee, John (Sydney) [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, June 13, 2008, 1:22 AM 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 ___ -- 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
Dear Jared; Any web in english? From where I can buy it? Regards Bilal -- 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
Re: [asterisk-users] Asterisk on SLOW solid state disk
On Wed, 11 Jun 2008, OCG Technical Support 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. M bits/sec or bytes/sec? If bytes, then that's a fast device! If bits, then it's about right. Has anyone tried a slow disk like this on asterisk? Will this delay voice prompts or screw up ast/linux in any interesting way? The easy answer is to not run directly off the flash, but to unload the flash into RAM and run from a ramdisk. This is what I do in my systems - boot off flash into RAM, then everything runs in RAM. Except a separate partition for voicemail. Eg: FilesystemSize Used Avail Use% Mounted on /dev/ram0 136M 105M 32M 77% / tmpfs 244M 0 244M 0% /dev/shm /dev/hda3 64M 1.9M 63M 3% /data Even if you're running live out of flash, it'll be fine as Linux will buffer everything up in RAM anyway, so you might have a 'hit' the first time round (unlikely though), but after that it ought to stay in RAM if you've got enough. (I know there are linux distros and Asterisk projects designed to run off CF, but I'm hoping to stay mainstream) I'd suggest rolling your own rather than running directly off flash though. Gordon ___ -- 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 schreef: Dear Jared; Any web in english? translate.google.com? Ron From where I can buy it? Regards Bilal -- 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 -- NeoNova BV, The Netherlands Professional internet and VoIP solutions http://www.neonova.nl Kruislaan 419 1098 VA Amsterdam info: 020-5628292 servicedesk: 020-5628292 fax: 020-5628291 KvK Amsterdam 34151241 The following disclaimer applies to this email: http://www.neonova.nl/maildisclaimer ___ -- 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] World Cheapest Predictive Dialer!
Hello, This looks an awful lot like an advertisement for a commercial product, which is only allowed on the biz list. Which you already posted this message to this week. I'm kind of confused. How do you get cheaper than free? Are you paying people to use your dialer? One other thing, it is illegal to scrub leads for a company against the USA FTC DNC lists unless those companies have paid the FTC and registered to have access to those leads, do you verify FTC registration before offering this service? MATT--- On 6/13/08, Muhammad Zulqarnain [EMAIL PROTECTED] wrote: Dear User! Although this email is intend for asterisk-business list however this might be useful for asterisk-user as well. Global IT Vision is proud to announce the World Cheapest Predictive Dialer. TeleRep Performance Optimizer Predictive Dialer is Hosted Web Based solution for Call Centers (with FREE DNC Scrubbing for US) that works from any where in the world with virtually unlimited agents. It's a prepaid pay as you go service. You just pay for the calls you make as our system allows you to add your own TRUNK so you can make calls to anywhere in the world with your own terminator. Pricing are as low as 0.014c/minute plus you will also save hundreds of $$$ with free DNC Scrubbing by using our hosted service. FEATURES LIST: · Free DNC Scrubbing for US Call Centers · Web Based Live Administration · Distributed Virtual Call Center · Campaign Management · Campaign Start/Stop Scheduling · Multiple Campaigns at a time · Agent Login from home · Press 1 for Live Transfer · Support from 1-1000 users · No Minimum Commitment · Pricing as low as 0.014c/minute · Use Your Own Carrier · No Dedicated Hardware/Software Required · Free Phone/Email Support · Live up-to-minute statistics Before starting TeleRep Performance Optimizer Predictive Dialer Solutions, our team along with Global IT Telecom Ltd A British Company amassed 7 years of experience building first class, mission-critical voice and Internet applications for large and small corporate clients. Our solution resides in a Tier-1 data center and employs the latest in voice and Internet technology to ensure security, redundancy, and the highest quality of service. Please contact [EMAIL PROTECTED] for more details! Thanks Regards, Muhammad Zulqarnain Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] http://www.gitv.pk ___ -- 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] Behind NAT: source is fring software (SIP)
Hi All; My Asterisk is behind NAT with IP Address 192.168.0.2. I configued on my iPlanet router and port forwarding for 5060 (UDP) to be forwarded for 192.168.0.2 and I was able to let the fring softphone (SIP) to register on the asterisk. But when caller initiate call, the caller hear the destination but the destination does not hear the caller. I checked the RTP port range and I found it (1 - 2) and I forwarded it for the internal IP address 192.168.0.2 but the problem stayed!! I do not know what should I do more? What it could be the reason for the problem? What should I do on the router more? I am also thinking if the fring software could use UDP ports other than the range setted in the rtp.conf? Is it possible that source to use different port than the Asterisk RTP ports? Note: do I have to do port forwarding on my router for the RTP UDP ports, or it is enough to forward the 5060 UDP port for the internal IP address 192.168.0.2? Any help? Regards Bilal ___ -- 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, How difficult would it be to have a JabberReceive Event *initiate* a channel ? I think that could be done. And you could also place Originate commands over AMI, as you mentioned it. You might be interested in BJ's work, as it covers that topic : http://www.asterisk.org/node/48440 Cheers, Philippe ___ -- 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.
Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 ___ -- 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] World Cheapest Predictive Dialer!
My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, This looks an awful lot like an advertisement for a commercial product, which is only allowed on the biz list. Which you already posted this message to this week. I'm kind of confused. How do you get cheaper than free? Are you paying people to use your dialer? One other thing, it is illegal to scrub leads for a company against the USA FTC DNC lists unless those companies have paid the FTC and registered to have access to those leads, do you verify FTC registration before offering this service? MATT--- On 6/13/08, Muhammad Zulqarnain [EMAIL PROTECTED] wrote: Dear User! Although this email is intend for asterisk-business list however this might be useful for asterisk-user as well. Global IT Vision is proud to announce the World Cheapest Predictive Dialer. TeleRep Performance Optimizer Predictive Dialer is Hosted Web Based solution for Call Centers (with FREE DNC Scrubbing for US) that works from any where in the world with virtually unlimited agents. It's a prepaid pay as you go service. You just pay for the calls you make as our system allows you to add your own TRUNK so you can make calls to anywhere in the world with your own terminator. Pricing are as low as 0.014c/minute plus you will also save hundreds of $$$ with free DNC Scrubbing by using our hosted service. FEATURES LIST: · Free DNC Scrubbing for US Call Centers · Web Based Live Administration · Distributed Virtual Call Center · Campaign Management · Campaign Start/Stop Scheduling · Multiple Campaigns at a time · Agent Login from home · Press 1 for Live Transfer · Support from 1-1000 users · No Minimum Commitment · Pricing as low as 0.014c/minute · Use Your Own Carrier · No Dedicated Hardware/Software Required · Free Phone/Email Support · Live up-to-minute statistics Before starting TeleRep Performance Optimizer Predictive Dialer Solutions, our team along with Global IT Telecom Ltd A British Company amassed 7 years of experience building first class, mission-critical voice and Internet applications for large and small corporate clients. Our solution resides in a Tier-1 data center and employs the latest in voice and Internet technology to ensure security, redundancy, and the highest quality of service. Please contact [EMAIL PROTECTED] for more details! Thanks Regards, Muhammad Zulqarnain Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] http://www.gitv.pk ___ -- 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.
Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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 ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Behind NAT: source is fring software (SIP)
Bilal, where are you? I bought an Allnet from an importer here in France. Otherwise, Germany or the UK will have them. Somewhere around 100 eu. randy ___ -- 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] Behind NAT: source is fring software (SIP)
Dear Randy; I am in Kuwait. From where I can buy it? Regards Bilal --- On Fri, 6/13/08, randulo [EMAIL PROTECTED] wrote: From: randulo [EMAIL PROTECTED] Subject: Re: [asterisk-users] Behind NAT: source is fring software (SIP) To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, June 13, 2008, 7:45 AM Bilal, where are you? I bought an Allnet from an importer here in France. Otherwise, Germany or the UK will have them. Somewhere around 100 eu. randy ___ -- 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, Sure, although I would have loved to see a pre-config dialplan:. Thanks for the tip. I think it will help me through. Best Regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, June 13, 2008 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: 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
Re: [asterisk-users] World Cheapest Predictive Dialer!
Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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] World Cheapest Predictive Dialer!
Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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] Astricon question: four or five tracks?
John Todd wrote: Is it too much to have 5 talk tracks at Astricon? Do the extra tracks. With a recording to review at night or online that nullifies the problem of picking. Really, with most presentations having slides all you need is fair video but excellent audio. How quick could this be turned around? In addition can you extend the hours of the vendor area. Last year it closed almost right after the talks. You had to pick between the talks and seeing what was new. ___ -- 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 Friday 13 June 2008 02:35:09 Nhadie Ramos wrote: Hi, I don't know what i'm doing wrong but i already reinstalled the system. still using ubuntu 64-bit. made sure i had the correct local date time. then did all this: ntpdate pool.ntp.org tzselect , i chose Asia/SIngapore /etc/timezone is Asia/Singapore i added TZ='Asia/Singapore'; export TZ to /etc/profile. date shows the correct date, i rebooted, date still shows the correct date. then installed zaptel, ./configure, make menuselect, make make install make config then libpri make make install then asterisk, ./configure, make menuselect, make, make install, make samples then asterisk-addons, ./configure make menuselect, make make install make samples then run asterisk, then connect via asterisk -r [Jun 13 00:25:58] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found [Jun 13 00:26:02] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found log shows June 13 00:25 my system date shows /home/ronald# date Fri Jun 13 15:33:03 SGT 2008 i installed everything as root via sudo su, how come i still dont get the correct time? The only thing I can think of is that your zoneinfo files are not in the right place. Does the file /usr/share/zoneinfo/Asia/Singapore exist? Also, does a symlink exist from that file to /etc/localtime? -- 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] World Cheapest Predictive Dialer!
I suppose if they are properly scrubbing (not the legal definition, but the practical definition of removing people that are on the DNC lists), then who is going to complain? Thanks, Steve T On Fri, Jun 13, 2008 at 8:19 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 10:51 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: 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 :) You're welcome :) Oh, btw, you will definitely need to enable unanswered = yes in cdr.conf as after ResetCDR new entry has disposition NO ANSWER, even if call is answered before. So without this you could loose them. 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
Hi Edgar,
Thanks for the reply. This setting is good for 10 simultaneous calls.
What i really need is 10 calls being done per second but no limit on
simultaneous calls.
On Fri, Jun 13, 2008 at 2:43 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
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 belowthe 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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VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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APUG! | http://asterisk.org.ph
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Re: [asterisk-users] World Cheapest Predictive Dialer!
A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless of the purpose of the calls. Cheers, Dean ___ -- 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
Steven Howes wrote: 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 LOL :) -- 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] Really destroying SIP dialog
c james wrote:
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.
c james,
Your best option would be to address the source of the messages, but I
know that's not always practical. Here is a trivial patch that will
only print the messages if verbosity is set to greater than 10. Just
apply it to 'channels/chan_sip.c' and rebuild Asterisk.
=== BEGIN PATCH
--- chan_sip.c 2008-06-13 08:51:46.0 -0400
+++ chan_sip.c.patched 2008-06-13 08:56:37.0 -0400
@@ -3115,7 +3115,8 @@
struct sip_pkt *cp;
if (sip_debug_test_pvt(p) || option_debug 2)
- ast_verbose(Really destroying SIP dialog '%s' Method:
%s\n, p-callid, sip_methods[p-method].text);
+ if (option_verbose 10)
+ ast_verbose(VERBOSE_PREFIX_4 Really destroying
SIP dialog '%s' Method: %s\n, p-callid, sip_methods[p-method].text);
if (ast_test_flag(p-flags[0], SIP_INC_COUNT) ||
ast_test_flag(p-flags[1], SIP_PAGE2_CALL_ONHOLD)) {
update_call_counter(p, DEC_CALL_LIMIT);
=== END PATCH ==
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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Re: [asterisk-users] asterisk calls per second
If you use .call files the you could write a script to create and mv
the .call files in batches of ten every second.
Maybe if you explain the purpose, someone might take more time to
think about it.
Thanks,
Steve T
On Fri, Jun 13, 2008 at 8:57 AM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
Hi Edgar,
Thanks for the reply. This setting is good for 10 simultaneous calls.
What i really need is 10 calls being done per second but no limit on
simultaneous calls.
On Fri, Jun 13, 2008 at 2:43 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
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 belowthe 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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Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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Re: [asterisk-users] World Cheapest Predictive Dialer!
Not sure who complains, but it has happened before. the first case was in 2006 when Phase One Marketing who was fined by the FTC for indirectly acquiring the FTC DNC list from another entity. MATT--- On 6/13/08, Steve Totaro [EMAIL PROTECTED] wrote: I suppose if they are properly scrubbing (not the legal definition, but the practical definition of removing people that are on the DNC lists), then who is going to complain? Thanks, Steve T On Fri, Jun 13, 2008 at 8:19 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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 ___ -- 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] World Cheapest Predictive Dialer!
Probably a whistle blower, disgruntled employee, or competitor. Thanks, Steve T On Fri, Jun 13, 2008 at 9:08 AM, Matt Florell [EMAIL PROTECTED] wrote: Not sure who complains, but it has happened before. the first case was in 2006 when Phase One Marketing who was fined by the FTC for indirectly acquiring the FTC DNC list from another entity. MATT--- On 6/13/08, Steve Totaro [EMAIL PROTECTED] wrote: I suppose if they are properly scrubbing (not the legal definition, but the practical definition of removing people that are on the DNC lists), then who is going to complain? Thanks, Steve T On Fri, Jun 13, 2008 at 8:19 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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 ___ -- 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] World Cheapest Predictive Dialer!
You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called Ameriquest was fined $1 million for violation of the DNC through it's affiliates, some of which were off-shore lead generation companies. The company shut down because of this fine. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless of the purpose of the calls. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
Hi,
I already gave a hint into right direction, but seems that it got
missed, so basically it would look like this:
exten=_3XX,1,Set(GROUP()=${EPOCH})
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(${EPOCH})})
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
Epoch is UNIX timestamp, which changes every second. Probably you
don't even need to use GROUP, but can keep counter for current second
in some database, however that would need database cleanups and locks.
Asterisk builtin DB wouldn't be useful, as it can't increment within
same operation, so some sort of SQL magic should be used. For example
multiple primary keys, one of which is autoincrement, or just
transactions.
However advantage of using GROUP would be that if call disconnects,
it's not counted within GROUP_COUNT anymore, so you can accept one
more call for that second(probably most useful for minute).
Regards,
Atis
On Fri, Jun 13, 2008 at 3:57 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
Hi Edgar,
Thanks for the reply. This setting is good for 10 simultaneous calls.
What i really need is 10 calls being done per second but no limit on
simultaneous calls.
On Fri, Jun 13, 2008 at 2:43 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
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 belowthe 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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Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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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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[asterisk-users] start n run an agi script on hangup
hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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] World Cheapest Predictive Dialer!
That can be avoided by simply distancing yourself through various corporate shell games. That's how the big boys do it. A good corporate lawyer can advise how to do this, but basically you setup a corporation that has no real assets that does business with the overseas company directly. Then you setup another totally separate corporation that uses the first corporation strictly as a vendor. Let them fine and and subsequently bankrupt the first corporation, with no assets, it is hard to get blood from a stone. Then the second corporation just needs to find a new vendor. It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Thanks, Steve T On Fri, Jun 13, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called Ameriquest was fined $1 million for violation of the DNC through it's affiliates, some of which were off-shore lead generation companies. The company shut down because of this fine. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless of the purpose of the calls. Cheers, Dean ___ -- 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] start n run an agi script on hangup
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI On Fri, Jun 13, 2008 at 9:29 AM, Robor Oghene [EMAIL PROTECTED] wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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] Any idea how making Asterisk transparent?
On Fri, Dec 7, 2007 at 6:43 AM, dave cantera [EMAIL PROTECTED] wrote: artifex, if you want call recording transparently, check out orecX.com they have a commercial and an open source SIP call recording package... no zap recording If you are using sangoma hardwarde it's possible to do a voice RTP tap for OrecX http://wiki.sangoma.com/wanpipe-voice-rtp-tap -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- 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] start n run an agi script on hangup
Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- 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] World Cheapest Predictive Dialer!
If that is the way you NEED to set things up then you are obviously a scumbag. (No referances to anyone on this list). If you start off with so many layers of shells, you obviously don't care what anyone thinks of you or your 'affiliated' companies. The laws were made to be pretty simple to follow, the lists can be expensive, but, IRC they are free up to 5 Area Codes. Yes the process to scrub may be a pain, but who wants to pitch to someone that DOESN'T want to be pitched to? I personally feel, and this is only an opinion, that you should follow the laws of the country to whom you are selling, calling, pitching. If there is a conflict in the law between source and target contries, it is better to follow the rules that are more strict. I lived for 4 years in a building where the Colombian Ambassador lived, he lived right above me, every Wednesday night starting at 11:00PM, he would have a party, lots of dancing on his hardwood floors, loud music, talking, banging, etc I went the first few times and asked him to please turn it down a notch as my kid needed to sleep for school in the morning, He NEVER complied, the Miami Police were called every Wednesday by the building security but were unable to make any arrests, or enact any type of authority because he was a diplomat and by extension his home was not governed by the laws of this country. In the end, I bribed the security into letting me into the Meter Room and after removing his US power meter, and having him stay in the dark until the power company could come the next morning, he learned to get along with others. This will happen to the Off-shore call centers that do not follow the rules, they will simply be forced to comply. It is not that hard to get a Valid DID from your ITSP that you can use to identify your outgoing calls, you can track call backs, both good and bad, Have it go to a VM box and allow someone to leave a message. If they are interested in being a customer, you gave them a way to reach you, if they are upset because you called, take them off your list. It is easy, use technology to save your workforce from un-needed work. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, June 13, 2008 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! That can be avoided by simply distancing yourself through various corporate shell games. That's how the big boys do it. A good corporate lawyer can advise how to do this, but basically you setup a corporation that has no real assets that does business with the overseas company directly. Then you setup another totally separate corporation that uses the first corporation strictly as a vendor. Let them fine and and subsequently bankrupt the first corporation, with no assets, it is hard to get blood from a stone. Then the second corporation just needs to find a new vendor. It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Thanks, Steve T On Fri, Jun 13, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called Ameriquest was fined $1 million for violation of the DNC through it's affiliates, some of which were off-shore lead generation companies. The company shut down because of this fine. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless
Re: [asterisk-users] World Cheapest Predictive Dialer!
Steve Totaro wrote: It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. I'm quite certain this is already obvious and will simply be interpreted as a tautological affirmation of the obvious, but such co-mingling of personal and business assets -- whether with an evidently fraudalent purpose or not as such -- will generally not survive the test of reasonableness that must be satisfied for corporate liability to not be pierced. In other words, if you simply pay for your house in this manner, and then you declare bankruptcy or are sued by creditors or whatever, the courts will scavenge this sort of thing up as evidence that your corporate entity is a financial alter-ego to whatever degree, and declare that your house is actually, de facto, a personal asset and can be included in the asset classes potentially awarded by judgments to the plaintiffs. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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, what i did is reinstall (again) but this time using debian 32-bit. and now i get the time correctly. so i'm not sure if it's a prob with ubuntu or asterisk, or asterisk on ubuntu, or asterisk on ubuntu 64-bit. coz i dont know how to figure those out. but anyway debian+asterisk works fine. thanks to all your reply! regards ron --- On Fri, 6/13/08, 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: Friday, June 13, 2008, 12:31 PM On Friday 13 June 2008 02:35:09 Nhadie Ramos wrote: Hi, I don't know what i'm doing wrong but i already reinstalled the system. still using ubuntu 64-bit. made sure i had the correct local date time. then did all this: ntpdate pool.ntp.org tzselect , i chose Asia/SIngapore /etc/timezone is Asia/Singapore i added TZ='Asia/Singapore'; export TZ to /etc/profile. date shows the correct date, i rebooted, date still shows the correct date. then installed zaptel, ./configure, make menuselect, make make install make config then libpri make make install then asterisk, ./configure, make menuselect, make, make install, make samples then asterisk-addons, ./configure make menuselect, make make install make samples then run asterisk, then connect via asterisk -r [Jun 13 00:25:58] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found [Jun 13 00:26:02] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found log shows June 13 00:25 my system date shows /home/ronald# date Fri Jun 13 15:33:03 SGT 2008 i installed everything as root via sudo su, how come i still dont get the correct time? The only thing I can think of is that your zoneinfo files are not in the right place. Does the file /usr/share/zoneinfo/Asia/Singapore exist? Also, does a symlink exist from that file to /etc/localtime? -- 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] Asterisk on SLOW solid state disk
25mb/sec isn't too bad it depends on how busy the system is. You could place most read prompts in to a ramdisk, however, the Linux kernel will cache frequently read files anyways... Brian On Wed, Jun 11, 2008 at 7:23 PM, 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. Has anyone tried a slow disk like this on asterisk? Will this delay voice prompts or screw up ast/linux in any interesting way? (I know there are linux distros and Asterisk projects designed to run off CF, but I'm hoping to stay mainstream) Thanks, MD ___ -- 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 -- -- Brian McManus ___ -- 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 10:35 AM, Alex Balashov [EMAIL PROTECTED] wrote: Steve Totaro wrote: It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. I'm quite certain this is already obvious and will simply be interpreted as a tautological affirmation of the obvious, but such co-mingling of personal and business assets -- whether with an evidently fraudalent purpose or not as such -- will generally not survive the test of reasonableness that must be satisfied for corporate liability to not be pierced. In other words, if you simply pay for your house in this manner, and then you declare bankruptcy or are sued by creditors or whatever, the courts will scavenge this sort of thing up as evidence that your corporate entity is a financial alter-ego to whatever degree, and declare that your house is actually, de facto, a personal asset and can be included in the asset classes potentially awarded by judgments to the plaintiffs. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 It is a legitimate real estate company renting you a place to live. This asset protection tactic has been around for a very long time and is legit. Totally separate entities. 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 10:10 AM, Alexander Lopez [EMAIL PROTECTED] wrote: If that is the way you NEED to set things up then you are obviously a scumbag. (No referances to anyone on this list). If you start off with so many layers of shells, you obviously don't care what anyone thinks of you or your 'affiliated' companies. I am just telling you how the big boys play. Like it or not. The laws were made to be pretty simple to follow, the lists can be expensive, but, IRC they are free up to 5 Area Codes. Yes the process to scrub may be a pain, but who wants to pitch to someone that DOESN'T want to be pitched to? I personally feel, and this is only an opinion, that you should follow the laws of the country to whom you are selling, calling, pitching. If there is a conflict in the law between source and target contries, it is better to follow the rules that are more strict. Then why did you break the law by tampering with the municipality's electric meter. Had he needed power for some sort of emergency, you could very well be held responsible in court. Whether involuntary manslaughter or something less. I lived for 4 years in a building where the Colombian Ambassador lived, he lived right above me, every Wednesday night starting at 11:00PM, he would have a party, lots of dancing on his hardwood floors, loud music, talking, banging, etc I went the first few times and asked him to please turn it down a notch as my kid needed to sleep for school in the morning, He NEVER complied, the Miami Police were called every Wednesday by the building security but were unable to make any arrests, or enact any type of authority because he was a diplomat and by extension his home was not governed by the laws of this country. In the end, I bribed the security into letting me into the Meter Room and after removing his US power meter, and having him stay in the dark until the power company could come the next morning, he learned to get along with others. See above, you are the one breaking the law. Using corporations for protection is why they were created. This is why the big boys use these laws to protect themselves and their assets, all legal like. This will happen to the Off-shore call centers that do not follow the rules, they will simply be forced to comply. By whom? The World Police? It is not that hard to get a Valid DID from your ITSP that you can use to identify your outgoing calls, you can track call backs, both good and bad, Have it go to a VM box and allow someone to leave a message. If they are interested in being a customer, you gave them a way to reach you, if they are upset because you called, take them off your list. It is easy, use technology to save your workforce from un-needed work. OK. Alex Thanks, Steve T -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, June 13, 2008 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! That can be avoided by simply distancing yourself through various corporate shell games. That's how the big boys do it. A good corporate lawyer can advise how to do this, but basically you setup a corporation that has no real assets that does business with the overseas company directly. Then you setup another totally separate corporation that uses the first corporation strictly as a vendor. Let them fine and and subsequently bankrupt the first corporation, with no assets, it is hard to get blood from a stone. Then the second corporation just needs to find a new vendor. It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Thanks, Steve T On Fri, Jun 13, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called
Re: [asterisk-users] Securing Asterisk and your network
On Thu, Jun 12, 2008 at 11:09:43PM +0300, Tzafrir Cohen 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 . Sure; all in-band methods suffer from the possibility of becoming DoS vectors. And yes, the fact that sshd doesn't quote that argument as it drops it into the syslog, making it easier to see bogusness, is a bad thing. But those log lines wouldn't fool *me*. And if they fool your log analysis system, then it's regexes aren't written tightly enough. And, back on point, that particular sshblocker doesn't give a damn what sshd writes in the syslog. And, no, it's actually not another service listening. 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] Astricon question: four or five tracks?
On Thu, Jun 12, 2008 at 08:52:27PM -0400, Steve Totaro wrote: 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. It's probably worth looking at the history of other large national technical conventions like Usenix and NANOG; Usenix makes proceedings available on line for free, and NANOG, the actual video recordings of the talks. The customary appraisal seems to be that this doesn't significantly affect the number of paid attendees, because there are many worthwhile advantages to physically attending the conference which you don't get from merely viewing the panel sessions on tape. Doing it is, admittedly, a non-trivial exercise... but it's a lot less difficult now than it used to be. Worth considering (he says, knowing that he won't be able to talk the boss into sending him... :-) 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 07:55:14AM -0400, Dean Collins wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I don't see that it's pertinent. FTC *owns* that list of numbers, and they can put whatever restrictions on it they like; I would assume that the restriction is contractual, not statutory. 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 09:42:13AM -0400, Steve Totaro wrote: It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Yup. But it'll cost you: at least in Florida, if a corporation owns your home, you don't get the $25,000 homestead exemption on your property taxes... 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 12:00 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Fri, Jun 13, 2008 at 09:42:13AM -0400, Steve Totaro wrote: It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Yup. But it'll cost you: at least in Florida, if a corporation owns your home, you don't get the $25,000 homestead exemption on your property taxes... 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) People can't even sell their property in most of FLA. $6,000 insurance (if you can get it) a year in Boca for a town house. Nobody is buying unless it is one of those We Buy Houses guys. Now they are the real scumbags. 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 11:39:46AM -0400, Steve Totaro wrote: [ Alex: ] I'm quite certain this is already obvious and will simply be interpreted as a tautological affirmation of the obvious, but such co-mingling of personal and business assets -- whether with an evidently fraudalent purpose or not as such -- will generally not survive the test of reasonableness that must be satisfied for corporate liability to not be pierced. In other words, if you simply pay for your house in this manner, and then you declare bankruptcy or are sued by creditors or whatever, the courts will scavenge this sort of thing up as evidence that your corporate entity is a financial alter-ego to whatever degree, and declare that your house is actually, de facto, a personal asset and can be included in the asset classes potentially awarded by judgments to the plaintiffs. It is a legitimate real estate company renting you a place to live. This asset protection tactic has been around for a very long time and is legit. Totally separate entities. Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. 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 on SLOW solid state disk
Hi ! We are running our asterisk from a transcend ts8gifd25. The whole system, including the OS fit in this 8 gig disk. If you don't do any recording of calls, you don't need that much of speed. We have 20 or so SIP phones, a PRI trough a quad-port sangoma card, one other port is a pass-trough to a dial-up RAS, another port is a point-to-point T1 data link to our office. To date, we've handle a handfull of simultaniously calls without any performance degradation. Regards, 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. Has anyone tried a slow disk like this on asterisk? Will this delay voice prompts or screw up ast/linux in any interesting way? (I know there are linux distros and Asterisk projects designed to run off CF, but I'm hoping to stay mainstream) 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
Re: [asterisk-users] World Cheapest Predictive Dialer!
Yup. But it'll cost you: at least in Florida, if a corporation owns your home, you don't get the $25,000 homestead exemption on your property taxes... Don't forget that you aren't protected by the 3% limit on property values, doesn't matter much now, but it did when the house across the street sold for 2.5 times what yours cost. Steve, Insurance is just one of the problems, try paying 12,000 a year in property taxes for services that cost 1,000 anywhere else in the country. We talk about DNC, FTC, and others, the real crooks are already here, and we supposedly ELECTED them! Alex 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 ___ -- 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] World Cheapest Predictive Dialer!
Hi Dean, Could you please tell me the source of information for your 2nd paragraph? I'd like to read up more. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: June 13, 2008 7:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- 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] PRI crashing Asterisk
I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows.. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 025 P/F: 1 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter tbkey*CLI soft hangup Zap/2-1 Requested Hangup on channel 'Zap/2-1' [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2973 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2612 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:2764 q931_disconnect: call 32770 on channel 2 enters state 11 (Disconnect Request) [ 00 01 32 34 08 02 00 02 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 025 0: 0 N(R): 026 P: 0 9 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2969 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 -- Hungup 'Zap/2-1' -- Hungup 'IAX2/1002-8371' host*CLI Disconnected from Asterisk server dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. ___ -- 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] World Cheapest Predictive Dialer!
Snip wrote: If that is the way you NEED to set things up then you are obviously a scumbag. (No referances to anyone on this list). If you start off with so many layers of shells, you obviously don't care what anyone thinks of you or your 'affiliated' companies. I am just telling you how the big boys play. Like it or not. I am going back to my sandbox!! The laws were made to be pretty simple to follow, the lists can be SNIP better to follow the rules that are more strict. Then why did you break the law by tampering with the municipality's electric meter. Had he needed power for some sort of emergency, you could very well be held responsible in court. Whether involuntary manslaughter or something less. All of the Life Safety Equipment was operational, it is powered by the building. He even had hot water and phone. Elevators, Fire Equipment, and hall lights are usually backed up via generator (at least here in Florida) That night we told him that there was a building power outage, We all turned off our lights and went to sleep. We delt with him in the morning when he found out he was the only one out.. I lived for 4 years in a building where the Colombian Ambassador lived, he lived right above me, every Wednesday night starting at 11:00PM, he SNIP US power meter, and having him stay in the dark until the power company could come the next morning, he learned to get along with others. See above, you are the one breaking the law. Using corporations for protection is why they were created. This is why the big boys use these laws to protect themselves and their assets, all legal like. There is always a fine line between Moraly right and Legally Right. I admit it when I pulled the meter, I was upset, it had been about 4 weeks of the same thing, we (Building Security and management, Police, Neighbors, and myself) tried to the best of our ability to resolve the problem. All we wanted was for a little respect for others. He felt he didn't need to comply. I have moved on, bought a single family house and I am much better off. Last I heard the Ambassador was sent back home. This will happen to the Off-shore call centers that do not follow the rules, they will simply be forced to comply. By whom? The World Police? The Consumer, nothing says change your ways more than no sales It is not that hard to get a Valid DID from your ITSP that you can use to identify your outgoing calls, you can track call backs, both good and bad, Have it go to a VM box and allow someone to leave a message. If they are interested in being a customer, you gave them a way to reach you, if they are upset because you called, take them off your list. It is easy, use technology to save your workforce from un-needed work. OK. Alex Thanks, Steve T SNIP ___ -- 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 make a ip to ip call
Hi, i want to make a direct ip to ip call (without a sip proxy), what software i can use (windows)? i try with xlite but dont understand how ___ -- 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 Fri, Jun 13, 2008 at 07:31:45AM -0500, Tilghman Lesher wrote: The only thing I can think of is that your zoneinfo files are not in the right place. Does the file /usr/share/zoneinfo/Asia/Singapore exist? Also, does a symlink exist from that file to /etc/localtime? This can be verified with 'zdump -v /etc/localtime' $ zdump -v /usr/share/zoneinfo/Asia/Singapore /usr/share/zoneinfo/Asia/Singapore Fri Dec 13 20:45:52 1901 UTC = Sat Dec 14 03:41:17 1901 SMT isdst=0 gmtoff=24925 /usr/share/zoneinfo/Asia/Singapore Sat Dec 14 20:45:52 1901 UTC = Sun Dec 15 03:41:17 1901 SMT isdst=0 gmtoff=24925 /usr/share/zoneinfo/Asia/Singapore Wed May 31 17:04:34 1905 UTC = Wed May 31 23:59:59 1905 SMT isdst=0 gmtoff=24925 /usr/share/zoneinfo/Asia/Singapore Wed May 31 17:04:35 1905 UTC = Thu Jun 1 00:04:35 1905 MALT isdst=0 gmtoff=25200 /usr/share/zoneinfo/Asia/Singapore Sat Dec 31 16:59:59 1932 UTC = Sat Dec 31 23:59:59 1932 MALT isdst=0 gmtoff=25200 /usr/share/zoneinfo/Asia/Singapore Sat Dec 31 17:00:00 1932 UTC = Sun Jan 1 00:20:00 1933 MALST isdst=1 gmtoff=26400 /usr/share/zoneinfo/Asia/Singapore Tue Dec 31 16:39:59 1935 UTC = Tue Dec 31 23:59:59 1935 MALST isdst=1 gmtoff=26400 /usr/share/zoneinfo/Asia/Singapore Tue Dec 31 16:40:00 1935 UTC = Wed Jan 1 00:00:00 1936 MALT isdst=0 gmtoff=26400 /usr/share/zoneinfo/Asia/Singapore Sun Aug 31 16:39:59 1941 UTC = Sun Aug 31 23:59:59 1941 MALT isdst=0 gmtoff=26400 /usr/share/zoneinfo/Asia/Singapore Sun Aug 31 16:40:00 1941 UTC = Mon Sep 1 00:10:00 1941 MALT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Sun Feb 15 16:29:59 1942 UTC = Sun Feb 15 23:59:59 1942 MALT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Sun Feb 15 16:30:00 1942 UTC = Mon Feb 16 01:30:00 1942 JST isdst=0 gmtoff=32400 /usr/share/zoneinfo/Asia/Singapore Tue Sep 11 14:59:59 1945 UTC = Tue Sep 11 23:59:59 1945 JST isdst=0 gmtoff=32400 /usr/share/zoneinfo/Asia/Singapore Tue Sep 11 15:00:00 1945 UTC = Tue Sep 11 22:30:00 1945 MALT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Sun Aug 8 16:29:59 1965 UTC = Sun Aug 8 23:59:59 1965 MALT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Sun Aug 8 16:30:00 1965 UTC = Mon Aug 9 00:00:00 1965 SGT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Thu Dec 31 16:29:59 1981 UTC = Thu Dec 31 23:59:59 1981 SGT isdst=0 gmtoff=27000 /usr/share/zoneinfo/Asia/Singapore Thu Dec 31 16:30:00 1981 UTC = Fri Jan 1 00:30:00 1982 SGT isdst=0 gmtoff=28800 /usr/share/zoneinfo/Asia/Singapore Mon Jan 18 03:14:07 2038 UTC = Mon Jan 18 11:14:07 2038 SGT isdst=0 gmtoff=28800 /usr/share/zoneinfo/Asia/Singapore Tue Jan 19 03:14:07 2038 UTC = Tue Jan 19 11:14:07 2038 SGT isdst=0 gmtoff=28800 -- 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] TCP UDP path not the same
HI, I got a one way audio when an ip phone dial to another ip phone in the same network. What I find is TCP UDP run different legs. Below is my configuration. asterisk (192.168.1.10) ipphone-A (192.168.1.111) ipphone-B (192.168.1.101) router (192.168.1.1) external IP (116.48.138.83) When A makes call to B, signal from A to router goes in the internal network. Then B pickup the call and I find that B will use external IP to reach the router. The signal from B finally can't reach to A. Below is a flow and you can see it involves using external IP. Is it related to the setting? Where and how to set it to make it work? U 192.168.1.10:5060 - 192.168.1.101:5060 INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0. Via: SIP/2.0/UDP 116.48.138.83:5060;branch=z9hG4bK612c1103;rport. From: 111 sip:[EMAIL PROTECTED];tag=as0a0b2a95. To: sip:[EMAIL PROTECTED]:5060. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: PBX. Max-Forwards: 70. Date: Fri, 13 Jun 2008 17:20:14 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 21200 21200 IN IP4 116.48.138.83. s=session. c=IN IP4 116.48.138.83. t=0 0. m=audio 19770 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. ___ -- 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 Fri, Jun 13, 2008 at 11:51:35AM -0400, Jay R. Ashworth wrote: On Thu, Jun 12, 2008 at 11:09:43PM +0300, Tzafrir Cohen 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 . Sure; all in-band methods suffer from the possibility of becoming DoS vectors. And yes, the fact that sshd doesn't quote that argument as it drops it into the syslog, making it easier to see bogusness, is a bad thing. But those log lines wouldn't fool *me*. And if they fool your log analysis system, then it's regexes aren't written tightly enough. Aparantly, getting the regex right is a bit trickier than people think. http://nvd.nist.gov/nvd.cfm?cvename=CVE-2007-4321 http://nvd.nist.gov/nvd.cfm?cvename=CVE-2006-6302 So getting this regex right is probably a bit tricky. And, back on point, that particular sshblocker doesn't give a damn what sshd writes in the syslog. And, no, it's actually not another service listening. It responds to external output. I can trigger it to run whenever I want. Pretty close to a service. Consider e.g. a spam filter used by a mail server. It might just as well have such remotely-exploitable security holes, if badly written. And the attacker does not even need direct access to the system running the spam filter. Or Asterisk handling proxied SIP/IAX traffic. -- 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
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 07:55:14AM -0400, Dean Collins wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. Another funny thing is how Internet infrastructure in the US is way cheaper than infrastructure in, say, Pakistan. That makes some companies use American providers such as Google and Microsoft for email and messaging. And host their web site in the US. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. It would become ironic rather than funny, if the company in question would think that it can do business with US companies on US infrastructure and yet annoy so many people in the US. -- 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] OT: Florida Property Taxes [was: World Cheapest Predictive Dialer!]
On Fri, Jun 13, 2008 at 12:38:44PM -0400, Alexander Lopez wrote: Steve, Insurance is just one of the problems, try paying 12,000 a year in property taxes for services that cost 1,000 anywhere else in the country. We talk about DNC, FTC, and others, the real crooks are already here, and we supposedly ELECTED them! The crooks are the people who passes Save-our-Homes. But that's offtopic for this list, certainly. 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] PRI crashing Asterisk
On Fri, Jun 13, 2008 at 12:52 PM, James Finstrom [EMAIL PROTECTED] wrote: I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows.. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 025 P/F: 1 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter tbkey*CLI soft hangup Zap/2-1 Requested Hangup on channel 'Zap/2-1' [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2973 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2612 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:2764 q931_disconnect: call 32770 on channel 2 enters state 11 (Disconnect Request) [ 00 01 32 34 08 02 00 02 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 025 0: 0 N(R): 026 P: 0 9 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2969 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 -- Hungup 'Zap/2-1' -- Hungup 'IAX2/1002-8371' host*CLI Disconnected from Asterisk server dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 Core dump? Try SIP instead of IAX2? 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] Securing Asterisk and your network
On Fri, Jun 13, 2008 at 08:43:44PM +0300, Tzafrir Cohen wrote: And if they fool your log analysis system, then it's regexes aren't written tightly enough. Aparantly, getting the regex right is a bit trickier than people think. http://nvd.nist.gov/nvd.cfm?cvename=CVE-2007-4321 http://nvd.nist.gov/nvd.cfm?cvename=CVE-2006-6302 So getting this regex right is probably a bit tricky. That can happen. And, back on point, that particular sshblocker doesn't give a damn what sshd writes in the syslog. And, no, it's actually not another service listening. It responds to external output. I can trigger it to run whenever I want. Pretty close to a service. Except that it's invisible to the outside world; it's a side-effect of sshd, without even it's own port. Consider e.g. a spam filter used by a mail server. It might just as well have such remotely-exploitable security holes, if badly written. And the attacker does not even need direct access to the system running the spam filter. Or Asterisk handling proxied SIP/IAX traffic. Sure, in general, being very particular about the taintedness of your data is an important security practice... 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
[asterisk-users] ASTERISK MFC R2 EWSD
Dear Dini Handayani, I read, you have installed Asterisk with Digium card TE110P , install MFC R2 connect to PSTN (indonesia) using DIG13 MFCR2 siemens EWSD, Germany. asterisk working normaly, outgoing call ok, incoming call ok. This mail Mon, 28 Apr 2008 05:51:17 Count you sent the configuration to siemens EWSD (trunk, MFC R2 etc) and the configuration Asterisk?, My Asterisk with card D110P Open Vox connect to siemens EWSD TECO Argentina, The Asterisk present problem for working in outgoing call and inconming call Thanks, Regards Mariano ___ -- 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] Odd Polycom Reboot Issue
Hi Tim, I'm not sure if this is the cause of your issue but we have found that all the polycoms with software 3.0.0 and below reboots upon receiving a call if there are multiple identical incoming INVITE requests in a sequence. This condition takes place when we use SER 0.9.7 and have multiple polycom phones subscribed and registered to the same number. You might want to listen to the SIP traffic to see if there are redundant INVITE requests coming in to the phone. The upcoming firmware 3.0.2 should fix this particular crash. Tim Nelson wrote: 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 ___ -- 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] World Cheapest Predictive Dialer!
I'm certainly learning a lot from this thread, especially from Steve Totaro. If only this was OT, I'd love to see a big fat discussion go on regarding this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 13, 2008 9:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! That can be avoided by simply distancing yourself through various corporate shell games. That's how the big boys do it. A good corporate lawyer can advise how to do this, but basically you setup a corporation that has no real assets that does business with the overseas company directly. Then you setup another totally separate corporation that uses the first corporation strictly as a vendor. Let them fine and and subsequently bankrupt the first corporation, with no assets, it is hard to get blood from a stone. Then the second corporation just needs to find a new vendor. It is similar to forming a corporation that owns your house and generates revenue from you paying rent (mortgage) payments. It is obviously a wash but your house is protected from any claims against you personally since it is owned by a total legally separate corporate entity. Thanks, Steve T On Fri, Jun 13, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called Ameriquest was fined $1 million for violation of the DNC through it's affiliates, some of which were off-shore lead generation companies. The company shut down because of this fine. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless of the purpose of the calls. Cheers, Dean ___ -- 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 ___ -- 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] Need a SIP trunk provider for US - Dallas/TX
All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can guarantee $500 minimum monthly spending. Until business ramps up, I doubt I'll be doing that kind of spending immediately. Suggestions? -- Dante ___ -- 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] Odd Polycom Reboot Issue
On Thu, Jun 12, 2008 at 4:32 PM, Tim Nelson [EMAIL PROTECTED] wrote: 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 Just getting down to basics but is this PoE or do you have a brick? Have you tried a different power brick or PoE port? 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] World Cheapest Predictive Dialer!
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, 13 June 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! On Fri, Jun 13, 2008 at 07:55:14AM -0400, Dean Collins wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. Another funny thing is how Internet infrastructure in the US is way cheaper than infrastructure in, say, Pakistan. That makes some companies use American providers such as Google and Microsoft for email and messaging. And host their web site in the US. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. It would become ironic rather than funny, if the company in question would think that it can do business with US companies on US infrastructure and yet annoy so many people in the US. No it's called playing by the rules of the game set by politicians and corporate leaders who set them. Just because you are a peon like me doesn't mean you cant play smarter than others. As way of example - check out this article; http://www.wired.com/techbiz/people/magazine/16-06/mf_hiroyuki?currentPa ge=1 I do love people who knows how to play the international hosting game.it's an aspect sorely lacking in most USA based web 2.0 people I've met. Cheers, Dean ___ -- 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] PRI crashing Asterisk
IAX2 wasjust the example for this output originating channel makes no difference. I can reproduce it zap to zap, sip to zap or iax2 to zap James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. On Fri, Jun 13, 2008 at 11:00 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Jun 13, 2008 at 12:52 PM, James Finstrom [EMAIL PROTECTED] wrote: I have a user who's system crashes on pri hangup request. Tried 1.4.19.1and 1.4.20 as well as the latest libpri no change Progress is as follows.. Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 025 P/F: 1 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter tbkey*CLI soft hangup Zap/2-1 Requested Hangup on channel 'Zap/2-1' [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2973 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/2-1 [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2612 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:2764 q931_disconnect: call 32770 on channel 2 enters state 11 (Disconnect Request) [ 00 01 32 34 08 02 00 02 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 025 0: 0 N(R): 026 P: 0 9 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [Jun 13 12:44:19] DEBUG[13420]: chan_zap.c:2969 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/2-1 -- Hungup 'Zap/2-1' -- Hungup 'IAX2/1002-8371' host*CLI Disconnected from Asterisk server dead... Thoughts? James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Quality and Toughness built in!! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 Core dump? Try SIP instead of IAX2? 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] IAX2 phones, BRI and Analogue cards
The site is x100p.com and I have bought the s100fx from them. It works fine for me. My intention was to use as a remote extension. Just be aware it will *NOT*accept domain name ONLY IP addresses. So if your server is a dynamic IP addresses you have use the public IP address and for some folks this may be a problem.. Support is only via an email. I've an HOWTO available if anyone is interested. On Thu, Jun 12, 2008 at 5:01 AM, Ade Vickers [EMAIL PROTECTED] wrote: 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 ___ -- 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] AEL Help
I need help translating extensions.conf to AEL:
[default]
exten = _X.,1,Set(DID=${EXTEN:6})
exten = _X.,n,Goto(continue,1)
exten = _1X.,1,Set(DID=${EXTEN:7})
exten = _1X.,n,Goto(continue,1)
exten = continue,1,Noop(${DID})
exten = continue,n,Set(GROUP(IAX)=incoming)
exten = continue,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail)
exten = continue,n,Goto(from-pri,${DID},1)
exten = continue,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
I need the above to goto AEL, here's what I have so far:
context default {
_X. = {
Set(DID=${EXTEN:6});
Goto(continue,1);
};
_1X. = {
Set(DID=${EXTEN:7});
Goto(continue,1);
};
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail);
Goto(from-pri,${DID},1);
fail:
Set(DIALSTATUS=CHANUNAVAIL);
};
};
My issue is I don't know what to do with the fail and continue goto statements.
Thanks.
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[asterisk-users] cdr-custom/Master.csv rotation
Hi, How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date? Thanks, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL Help
Jeremy Mann wrote:
I need help translating extensions.conf to AEL:
[default]
exten = _X.,1,Set(DID=${EXTEN:6})
exten = _X.,n,Goto(continue,1)
exten = _1X.,1,Set(DID=${EXTEN:7})
exten = _1X.,n,Goto(continue,1)
exten = continue,1,Noop(${DID})
exten = continue,n,Set(GROUP(IAX)=incoming)
exten = continue,n,GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail)
exten = continue,n,Goto(from-pri,${DID},1)
exten = continue,n(fail),Set(DIALSTATUS=CHANUNAVAIL)
I need the above to goto AEL, here's what I have so far:
context default {
_X. = {
Set(DID=${EXTEN:6});
Goto(continue,1);
};
_1X. = {
Set(DID=${EXTEN:7});
Goto(continue,1);
};
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
GotoIf($[${MATH(${GROUP_COUNT([EMAIL
PROTECTED])}+${GROUP_COUNT([EMAIL PROTECTED])},i)}10]?fail);
Goto(from-pri,${DID},1);
fail:
Set(DIALSTATUS=CHANUNAVAIL);
};
};
My issue is I don't know what to do with the fail and continue goto
statements.
Thanks.
This e-mail, facsimile, or letter and any files or attachments transmitted
with it contains information that is confidential and privileged. This
information is intended only for the use of the individual(s) and entity(ies)
to whom it is addressed. If you are the intended recipient, further
disclosures are prohibited without proper authorization. If you are not the
intended recipient, any disclosure, copying, printing, or use of this
information is strictly prohibited and possibly a violation of federal or
state law and regulations. If you have received this information in error,
please notify Texas Health Management Group immediately at 1-817-310-4999.
Texas Health Management Group, its subsidiaries, and affiliates hereby claim
all applicable privileges related to this information.
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I'll be more than glad to help :)
Here's the code:
context default {
_X. = {
Set(DID=${EXTEN:6});
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
if(${MATH(${GROUP_COUNT([EMAIL PROTECTED])}+${GROUP_COUNT([EMAIL
PROTECTED])},i)}
10) {
Set(DIALSTATUS=CHANUNAVAIL);
}
jump [EMAIL PROTECTED];
}
}
You didn't really need the continue or fail label, as the code that you
had at fail is taken care of within the if statement's execution.
Let me know if there's any issue, if there is it's probably in the
implementation of the conditional
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] Odd Polycom Reboot Issue
Well... the problem magically cleared itself up this morning. Nothing was changed in the meantime. I'd love to know what the problem was... Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 13, 2008 1:31:36 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Odd Polycom Reboot Issue On Thu, Jun 12, 2008 at 4:32 PM, Tim Nelson [EMAIL PROTECTED] wrote: 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 Just getting down to basics but is this PoE or do you have a brick? Have you tried a different power brick or PoE port? 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] AEL Help
Sherwood McGowan wrote:
snip
I'll be more than glad to help :)
Here's the code:
context default {
_X. = {
Set(DID=${EXTEN:6});
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
if(${MATH(${GROUP_COUNT([EMAIL PROTECTED])}+${GROUP_COUNT([EMAIL
PROTECTED])},i)}
10) {
Set(DIALSTATUS=CHANUNAVAIL);
}
jump [EMAIL PROTECTED];
}
}
You didn't really need the continue or fail label, as the code that
you had at fail is taken care of within the if statement's execution.
Let me know if there's any issue, if there is it's probably in the
implementation of the conditional
Ooops, remove the continue: line, it's not needed
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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Re: [asterisk-users] start n run an agi script on hangup
Thanks Steve, I appreciate your response, I checked the link and it talks about an agi script running before and continuing after hangup. the problem I have is that, I dont want to run an agi while the channel is up. i want to start the script on on hangup to do database cleanup.. i'd appreciate if you'd shed more light just in case am missing something.. Rgds On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- 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 ___ -- 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] start n run an agi script on hangup
Robor Oghene wrote: Thanks Steve, I appreciate your response, I checked the link and it talks about an agi script running before and continuing after hangup. the problem I have is that, I dont want to run an agi while the channel is up. i want to start the script on on hangup to do database cleanup.. i'd appreciate if you'd shed more light just in case am missing something.. Rgds On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] mailto:[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 ___ -- 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 DeadAGI will run even after both channels are down, that's why they called it DeadAGI. VERY useful when put in the h exten :) -- 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] start n run an agi script on hangup
Thanks Sherwood, I haven't tried what the line you sent but I think it would solve my problem I just hope it would run from asterisk realtime... On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- 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 ___ -- 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.
2008/6/12 Syed Nasruddin [EMAIL PROTECTED]: 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. Our clients use this for E1 Pri: http://www.voicetronix.com/logger.htm Not sure if there is a analogue solution. -- http://www.suretecsystems.com/services/openldap/ ___ -- 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] cdr-custom/Master.csv rotation
2008/6/13 Mark Hamilton [EMAIL PROTECTED]: Hi, How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date? Logrotate on a *nix box. -- http://www.suretecsystems.com/services/openldap/ ___ -- 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] Need a SIP trunk provider for US - Dallas/TX
I use bandwidth.com, works very well. 5 trunks start at about $125 a month. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D. Dante Lorenso Sent: Friday, June 13, 2008 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Need a SIP trunk provider for US - Dallas/TX All, I'm in Dallas, TX, US and am looking for inbound-only DID service with 10+ channels on a SIP trunk. Is anyone on this list doing something similar and have any recommendations for a provider? Of course I'll be routing either SIP/IAX to an asterisk server that will be hosted in a Dallas colocation facility. I found VoxBone.com but they only want to deal with customers that can guarantee $500 minimum monthly spending. Until business ramps up, I doubt I'll be doing that kind of spending immediately. Suggestions? -- Dante ___ -- 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] start n run an agi script on hangup
Thanks a million Sherwood!! On Fri, Jun 13, 2008 at 9:31 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Robor Oghene wrote: Thanks Steve, I appreciate your response, I checked the link and it talks about an agi script running before and continuing after hangup. the problem I have is that, I dont want to run an agi while the channel is up. i want to start the script on on hangup to do database cleanup.. i'd appreciate if you'd shed more light just in case am missing something.. Rgds On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] mailto:[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 ___ -- 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 DeadAGI will run even after both channels are down, that's why they called it DeadAGI. VERY useful when put in the h exten :) -- 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 ___ -- 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] World Cheapest Predictive Dialer!
On Fri, Jun 13, 2008 at 02:37:52PM -0400, Dean Collins wrote: No it's called playing by the rules of the game set by politicians and corporate leaders who set them. The same politicials who set those nasty anti-spam laws? -- 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
Re: [asterisk-users] Odd Polycom Reboot Issue
Nevermind. The issue came back. In response to Steve's query... the unit is running over PoE. When time permits, I'll try a different port on the switch. I've been told that a change was made to the volume ringer of the phones in the sip.cfg to a value of 32 if this makes any difference. All suggestions welcome! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Tim Nelson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 13, 2008 2:54:25 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Odd Polycom Reboot Issue Well... the problem magically cleared itself up this morning. Nothing was changed in the meantime. I'd love to know what the problem was... Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 13, 2008 1:31:36 PM GMT -06:00 US/Canada Central Subject: Re: [asterisk-users] Odd Polycom Reboot Issue On Thu, Jun 12, 2008 at 4:32 PM, Tim Nelson [EMAIL PROTECTED] wrote: 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 Just getting down to basics but is this PoE or do you have a brick? Have you tried a different power brick or PoE port? 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
[asterisk-users] strange iax authentication behavior
Hi. I have a number of sections in my iax configurationwhich was generated by freepbx like this: [covici] username=covici type=friend secret=password host=dynamic disallow=all context=from-internal allow=ulaw [Carol 0326] host=iax.binfone.com username=3234740326 secret=another password type=friend context=from-pstn Now with this configuration, if I receive a call over the trunk covici, it complains that the host sending the call is trying to authenticate as Carol 0326 . If I change the second trunk to type=peer it works, or if I put the covici trunk at the end it also works -- anyone know why this is happening because of course the config gets regenerated all the time and this should not work this way. Thanks in advance for any assistance. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] Trouble with Polycom phones
No, even with the numerical IP addresses they still had the problem. Kevin Mike wrote: I`m curious: did going with numerical IP addresses fix your problem? Mick -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Smith Sent: Wednesday, June 04, 2008 13:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with Polycom phones Yes, I was using a name instead of an IP address. And if memory servesI *think* it is using TCPprefered...but I could be wrong. Kevin Mike wrote: I have been running into a few issues with Asterisk/polycom and I am running out of ideas. This problem has been ongoing for the last couple of weeks. I will try to be as detailed as I can, but I might leave out a few details. Any suggestions would be greatly appreciated. Now, the phones lose their registration with Asterisk. Are you using a numeric IP address or a name for the Asterisk server in the Polycom config? I had the same issue (only from 2.2 up IIRC) until I put in the numerical IP. Can't explain it, maybe somebody else can. Mick ___ -- 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 -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- 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] Idiot's question
I have two lines in my dialplan that I wish to make it into only one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the first 11
chars from the left. It aims to detect null caller ids and replace them by
zeros. How can I write this expression in just one line?
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Re: [asterisk-users] start n run an agi script on hangup
Robor Oghene wrote: Thanks Sherwood, I haven't tried what the line you sent but I think it would solve my problem I just hope it would run from asterisk realtime... On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] mailto:[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 ___ -- 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 You can't store AEL in realtime extensions, sorry, it's compiled at load time (or AEL reload + dialplan reload) by the compiler :( -- 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] start n run an agi script on hangup
Robor Oghene wrote: Thanks Sherwood, I haven't tried what the line you sent but I think it would solve my problem I just hope it would run from asterisk realtime... On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] mailto:[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 ___ -- 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 oops, sorry I was thinking of another post I had responded too :( Yes, DeadAGI and the h extension is handled just fine in realtime extensions -- 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] Idiot's question
Venefax schrieb:
I have two lines in my dialplan that I wish to make it into only one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the first 11
chars from the left. It aims to detect null caller ids and replace them by
zeros. How can I write this expression in just one line?
I think it does multiple passes to evaluate ${} so maybe
Set(CALLERID(num)=${${CALLERID(num)}00:0:11})
works.
However assuming a callerid to always be 11 chars is not
generally valid.
Grüße,
Philipp Kempgen
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Re: [asterisk-users] Idiot's question
On Fri, 13 Jun 2008, Venefax wrote:
I have two lines in my dialplan that I wish to make it into only one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the first 11
chars from the left. It aims to detect null caller ids and replace them by
zeros. How can I write this expression in just one line?
First, it doesn't do what your explanation says it does.
Second, why would you want to? Any savings in execution time will be
insignificant and it will obscure the intent and readability of the
code.
FYI, my 2.6GHz P4 executes about 3,000 CALLERID(num) assignments per
second.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Idiot's question
On Sat, 14 Jun 2008, Philipp Kempgen wrote:
Venefax schrieb:
I have two lines in my dialplan that I wish to make it into only one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the first 11
chars from the left. It aims to detect null caller ids and replace them by
zeros. How can I write this expression in just one line?
I think it does multiple passes to evaluate ${} so maybe
Set(CALLERID(num)=${${CALLERID(num)}00:0:11})
works.
Not in 1.2.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] start n run an agi script on hangup
Two things jump to mind depending on the nature of your script. Both use the h extension, this is where a call will go when hungup. You can either use deadagi on the h extension or you can use exten = h,1,System(command). Thanks, Steve T On Fri, Jun 13, 2008 at 4:26 PM, Robor Oghene [EMAIL PROTECTED] wrote: Thanks Steve, I appreciate your response, I checked the link and it talks about an agi script running before and continuing after hangup. the problem I have is that, I dont want to run an agi while the channel is up. i want to start the script on on hangup to do database cleanup.. i'd appreciate if you'd shed more light just in case am missing something.. Rgds On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- 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 ___ -- 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] World Cheapest Predictive Dialer!
Hi! First of all I apologize for posting this commercial email on asterisk-user list but it gave me some valuable information which were not notified by any one on asterisk-buzz list. Second I would like to clear the confusion created by my email about FREE DNC Scrubbing. Our Predictive Dialer have capability to filter list of number against the List of National Do Not Call Registry provided by the client itself. So they don't have to buy any additional software from market, like dnclistmanager, Scrub DNC etc. for scrubbing their list before dialing their leads. In this way let me know if even we from developing country like Pakistan are violating US or FTC laws. Some one asked in the list that how It is world most cheapest predictive dialer. We don't pay to people for using our software. TeleRep Performance Optimizer is not a free predictive dialer but the most cheapest Predictive Dialer (just 0.014c per minute, as you use your own carrier to terminate call anywhere in the world, but not only to USA) as compare to other Hosted Dialer available in the market like Callfire and many others available. However, discussion made by several friends including Steve, MATT, Dean are very valuable and I thanks all of you for taking your time on discussing this matter on list. I am looking forward further for your feedback on this matter if still it is out of law. Thanks Regards, Zulqarnain ___ -- 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] World Cheapest Predictive Dialer!
On Friday 13 June 2008 23:07:17 Muhammad Zulqarnain wrote: Some one asked in the list that how It is world most cheapest predictive dialer. We don't pay to people for using our software. TeleRep Performance Optimizer is not a free predictive dialer but the most cheapest Predictive Dialer (just 0.014c per minute, as you use your own carrier to terminate call anywhere in the world, but not only to USA) as compare to other Hosted Dialer available in the market like Callfire and many others available. Clearly, you missed the point. Since there is a FREE predictive dialer out there, and your product costs something, you are not the world's cheapest predictive dialer. The only way you could possibly be cheaper than free is if you paid people to use your product. Not a particularly wise business plan, but then, what do I know? -- 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] Idiot's question
You could always do something like this, but I think it may pull more
proc than on 2 lines:
exten = s,n,Set(CALLERID(num)=${IF($[foo${CALLERID(num)} = foo]?
00:${CALLERID(num)})})
On Jun 13, 2008, at 9:12 PM, Steve Edwards wrote:
On Sat, 14 Jun 2008, Philipp Kempgen wrote:
Venefax schrieb:
I have two lines in my dialplan that I wish to make it into only
one, and I
fail
X.,n(entrada),Set(CALLERID(num)=${CALLERID(num)}00)
X.,n,Set(CALLERID(num)=${CALLERID(num):0:11})
It means: add '00' to the caller id, and then take the
first 11
chars from the left. It aims to detect null caller ids and replace
them by
zeros. How can I write this expression in just one line?
I think it does multiple passes to evaluate ${} so maybe
Set(CALLERID(num)=${${CALLERID(num)}00:0:11})
works.
Not in 1.2.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
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