[asterisk-users] zaptel compile kernel problem
Hi guys, I am trying to compile zaptel, using debian 4r5. However what I get in zaptel 1.2.27 after make is below : You do not appear to have the sources for the 2.6.18-6-486 kernel installed (under ). make: *** [modules] Error 1 tried to change the source with zaptel-1.4.12.1 :/usr/src/zaptel-1.4.12.1# make make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo You do not appear to have the sources for the 2.6.18-6-486 kernel installed. You do not appear to have the sources for the 2.6.18-6-486 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' make: *** [all] Error 2 I think I do have that kerne, after inputting uname -r what I get is : asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 what should I do? thank you ___ -- 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] zaptel compile kernel problem
reza adinata wrote: Hi guys, I am trying to compile zaptel, using debian 4r5. However what I get in zaptel 1.2.27 after make is below : You do not appear to have the sources for the 2.6.18-6-486 kernel installed (under ). make: *** [modules] Error 1 tried to change the source with zaptel-1.4.12.1 :/usr/src/zaptel-1.4.12.1# make make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo You do not appear to have the sources for the 2.6.18-6-486 kernel installed. You do not appear to have the sources for the 2.6.18-6-486 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' make: *** [all] Error 2 I think I do have that kerne, after inputting uname -r what I get is : asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 what should I do? thank you Install the kernel source. DC ___ -- 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] zaptel compile kernel problem
asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? thank you On 2/17/09, Dave Cotton dcot...@linuxautrement.com wrote: reza adinata wrote: Hi guys, I am trying to compile zaptel, using debian 4r5. However what I get in zaptel 1.2.27 after make is below : You do not appear to have the sources for the 2.6.18-6-486 kernel installed (under ). make: *** [modules] Error 1 tried to change the source with zaptel-1.4.12.1 :/usr/src/zaptel-1.4.12.1# make make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo You do not appear to have the sources for the 2.6.18-6-486 kernel installed. You do not appear to have the sources for the 2.6.18-6-486 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12.1' make: *** [all] Error 2 I think I do have that kerne, after inputting uname -r what I get is : asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 what should I do? thank you Install the kernel source. DC ___ -- 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] freemin managment for sim cards
is any program , to manage freemin on sim cards ,for gsm gateways that connected to the asterisk, for termination? ___ -- 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] zaptel compile kernel problem
On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? aptitude install zaptel-source m-a prepare m-a build zaptel Those commands will not install that version of Zaptel, and hence you should not worry about version collisions. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] zaptel compile kernel problem
Hi, Yes, it is indeed working. I am currently using a debian4r5, and i can install using aptitude The problem is that I am trying to install an asterisk mp3player using mpg123 that is capable of playing from .pls. And in some literatures I have read, it is mentioned that I should have the newest asterisk, and the newest mpg123.. and a similar problem occured when I installed the newest asterisk using a source .. (inappropriate kernel) :( On 2/17/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? aptitude install zaptel-source m-a prepare m-a build zaptel Those commands will not install that version of Zaptel, and hence you should not worry about version collisions. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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 ___ -- 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] zaptel compile kernel problem
Tzafrir Cohen wrote: On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? Having said this to other people and received the follow up ~$ m-a command not found its nicer to say apt-get/aptitude install module assistant first. Bails aptitude install zaptel-source m-a prepare m-a build zaptel Those commands will not install that version of Zaptel, and hence you should not worry about version collisions. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] zaptel compile kernel problem
On Tue, Feb 17, 2009 at 09:54:21AM +, bails wrote: Tzafrir Cohen wrote: On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? Having said this to other people and received the follow up ~$ m-a command not found Which only goes to show you have not followed the instructions. http://packages.debian.org/etch/zaptel-source zaptel-source depends on module-assistant. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] zaptel compile kernel problem
On Tue, Feb 17, 2009 at 10:30:31AM +, Gordon Henderson wrote: On Tue, 17 Feb 2009, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 Just a minor issue here - there was an issue with kernels 2.6.18 whereby a user could get root access by running a simple program. I'm not sure if Debian patched it though, but it might be worthwhile checking and upgrading the kernel if neccessary. http://www.debian.org/security/2008/dsa-1494 (Feb 2008) Debian 4.0r5 was released at Oct 2008 (http://www.debian.org/News/2008/20081023) and included that fix. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] zaptel compile kernel problem
On Tue, Feb 17, 2009 at 05:19:42PM +0700, reza adinata wrote: i am sorry, but I am not using English as my main language.. A bit confused with several explanations above :( what i get is that : asterisk:/home/tsp# aptitude install zaptel-source aptitude not installed? Well, just use apt-get isntead apt-get install zaptel-source bash: aptitude: command not found asterisk:/home/tsp# m-a prepare bash: m-a: command not found asterisk:/home/tsp# m-a build zaptelsu bash: m-a: command not found As mentioned in a different message, m-a comes from a package that is a dependency of zaptel-source and hence should be available once you have that installed. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] zaptel compile kernel problem
i am sorry, but I am not using English as my main language.. A bit confused with several explanations above :( what i get is that : asterisk:/home/tsp# aptitude install zaptel-source bash: aptitude: command not found asterisk:/home/tsp# m-a prepare bash: m-a: command not found asterisk:/home/tsp# m-a build zaptelsu bash: m-a: command not found (getting really confusing right now) On 2/17/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Feb 17, 2009 at 09:54:21AM +, bails wrote: Tzafrir Cohen wrote: On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? Having said this to other people and received the follow up ~$ m-a command not found Which only goes to show you have not followed the instructions. http://packages.debian.org/etch/zaptel-source zaptel-source depends on module-assistant. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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 ___ -- 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] zaptel compile kernel problem
On Tue, Feb 17, 2009 at 04:48:37PM +0700, reza adinata wrote: On 2/17/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Feb 17, 2009 at 04:27:39PM +0700, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 doesn't that mean that I have already got the precise version in my box? (uname - r-kernel-release print the kernel release) ? why do I have to install the same kernel? Here's a sanity check for you: can your system build the module zaptel? aptitude install zaptel-source m-a prepare m-a build zaptel Those commands will not install that version of Zaptel, and hence you should not worry about version collisions. Yes, it is indeed working. I am currently using a debian4r5, and i can install using aptitude I have not asked about installing pre-built packages. I asked you to test-build the zaptel source from zaptel-source . If it builds, you should have working kernel headers. The problem is that I am trying to install an asterisk mp3player using mpg123 that is capable of playing from .pls. And in some literatures I have read, it is mentioned that I should have the newest asterisk, and the newest mpg123.. and a similar problem occured when I installed the newest asterisk using a source .. (inappropriate kernel) :( Is the problem lack of a timing source? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] What is the purpose of membermacro in queues.conf
Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj ___ -- 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] zaptel compile kernel problem
On Tue, 17 Feb 2009, reza adinata wrote: asterisk:/usr/src/zaptel-1.2.27# uname -r 2.6.18-6-486 Just a minor issue here - there was an issue with kernels 2.6.18 whereby a user could get root access by running a simple program. I'm not sure if Debian patched it though, but it might be worthwhile checking and upgrading the kernel if neccessary. 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
[asterisk-users] Pingable and Unreachable at the same time !
Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Pingable and Unreachable at the same time !
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Regards, Marc From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: mardi 17 février 2009 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196http://192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Credit Card processing machines
On Tue, 17 Feb 2009, Andrew Joakimsen wrote: On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote: Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look into reducing the baud rate on the machines, but sadly the bank has them password protected and wants to charge a reprogramming fee :( They make credit card terminals with Ethernet -- use that instead. The client's processor charges 7c/transaction over IP (plus normal charges), so they are quite keen to keep it working the way it was before I replaced their PBX ;) As a followup, *99 prepended on any Linksys ATA does indeed make a difference in modem reliability. Both their CCs and their ADT alarm devices now function reliably. I also reduced the CC baud rate to 300 baud (!), and it is rock solid now! j ___ -- 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] Pingable and Unreachable at the same time !
2009/2/17 Marc STORCK msto...@voipgate.com Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Yes. I think that simply, in this case, the endpoint (SIP phone) is just broken : it wouldn't reply to anything ... I'm not 100% sure now, but wouldn't be surprised ... Regards, Marc *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* mardi 17 février 2009 14:06 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Pingable and Unreachable at the same time !
Did you use the same screen name / name for the 2 SIP extensions you setup on the one phone? If so, some phones will confuse asterisk based on the SIP header (in particular AASTRA phones). If this is an Aastra phone, this is probably the cause... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: February 17, 2009 8:47 AM To: Asterisk Users List Subject: Re: [asterisk-users] Pingable and Unreachable at the same time ! 2009/2/17 Marc STORCK msto...@voipgate.com Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message. Yes. I think that simply, in this case, the endpoint (SIP phone) is just broken : it wouldn't reply to anything ... I'm not 100% sure now, but wouldn't be surprised ... Regards, Marc From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: mardi 17 février 2009 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pingable and Unreachable at the same time ! Hi, Has anyone met something like this ? dialor*CLI sip show peers Name/username HostDyn Nat ACL Port Status 7541/7541 (Unspecified)D 0UNKNOWN 7540/7540 (Unspecified)D 0UNKNOWN 7534/7534 (Unspecified)D 0UNKNOWN 7533/7533 (Unspecified)D 0UNKNOWN 7531/7531 192.168.100.199 D 5060 OK (10 ms) 7530/7530 192.168.100.196 D 5060 UNREACHABLE patton/patton 192.168.100.52 D 5060 OK (33 ms) trunk/trunk4ipbx 192.168.64.25060 OK (1 ms) 8 sip peers [Monitored: 3 online, 5 offline Unmonitored: 0 online, 0 offline] dialor*CLI !ping 192.168.100.196 PING 192.168.100.196 (192.168.100.196) 56(84) bytes of data. 64 bytes from 192.168.100.196: icmp_seq=1 ttl=64 time=0.334 ms 64 bytes from 192.168.100.196: icmp_seq=2 ttl=64 time=0.305 ms 64 bytes from 192.168.100.196: icmp_seq=3 ttl=64 time=0.305 ms Any explaination ? Regards ___ -- 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] Hangup extensions via CLI?
Ok isn't this replacing a western hack with a bridge hack? The init 0 and init 6 probably aren't going to work anyway since (1) asterisk has to be running as root and (2) the path in * is limited if even existent, so the init command would work unless you had a copy or symlink in the asterisk directory. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, February 15, 2009 11:26 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup extensions via CLI? On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote: This will hang-up all channels even if multiples channels are open... Exten = _86,1,system(init 0) Use with Caution.? Only if Asterisk is running as root. Which is not recommended, anyway. And besides, I think you meant: Exten = _86,1,system(init 6) as we want to leave the extension available afterwards. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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 ___ -- 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] Packet Truncated - Choppy Audio
Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654-8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232. Could it be that Asterisk is chopping the packets? Has anyone seen this before? Any assistance would be most gratefully received. Regards, Matt King Managing Director Orderly Software Ltd. ___ -- 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] Packet Truncated - Choppy Audio
This indicates that your NIC card is not handling the throughput effectively. Is * the only application on your server? How many users are on * when this occurs? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt King Sent: Tuesday, February 17, 2009 8:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Packet Truncated - Choppy Audio Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654-8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232. Could it be that Asterisk is chopping the packets? Has anyone seen this before? Any assistance would be most gratefully received. Regards, Matt King Managing Director Orderly Software Ltd. ___ -- 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] Network architecture
Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other stuff plus a MySQL database... This architecture convinced me, but I have some questions regarding asterisk and I need asterisk expert answers in order to take decision... 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? Waiting your reply Regards ___ -- 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] Network architecture
On Tue, 17 Feb 2009, michel freiha wrote: Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other stuff plus a MySQL database... This architecture convinced me, but I have some questions regarding asterisk and I need asterisk expert answers in order to take decision... 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? First off I think you would have a rough time making one server handle so many calls. It also depends heavily on whether or not you will be transcoding those calls. Regardless you should split the load for the simple reason that such a high density service would be in absolute tatters if your single point of failure failed for any reason. Are you hiring?? :) j ___ -- 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] Network architecture
No, asterisk on conventional hardware can handle at most a few hundred calls. I would strongly discourage the use of Asterisk purely as a transit element for billing. Just because a2billing is available does not mean you should. Far more scalable solutions are easily available. -- Sent from mobile device On Feb 17, 2009, at 10:19 AM, michel freiha mich...@gmail.com wrote: Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other stuff plus a MySQL database... This architecture convinced me, but I have some questions regarding asterisk and I need asterisk expert answers in order to take decision... 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? Waiting your reply Regards ___ -- 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] Network architecture
Just a laypersons opinion - I'm sure others here have better answers or justifications. 1. no (at least not realistically, mathematically there are some) 2. perhaps - bandwidth would be your primary concern since 5K calls would take 150 M of bandwidth 3. IMO it would be better to divide the load, but this depends on the hardware you are using. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Tuesday, February 17, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users-boun...@lists.digium.com Subject: [asterisk-users] Network architecture Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other stuff plus a MySQL database... This architecture convinced me, but I have some questions regarding asterisk and I need asterisk expert answers in order to take decision... 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? Waiting your reply Regards ___ -- 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] Network architecture
2009/2/17 Danny Nicholas da...@debsinc.com Just a laypersons opinion – I'm sure others here have better answers or justifications. 1. no (at least not realistically, mathematically there are some) 2. perhaps – bandwidth would be your primary concern since 5K calls would take 150 M of bandwidth 3. IMO it would be better to divide the load, but this depends on the hardware you are using. I would recommend opensips with cdrtool and mediaproxy all load balanced with heartbeat or dns. ___ -- 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] Hangup extensions via CLI?
On Tue, Feb 17, 2009 at 08:57:51AM -0600, Danny Nicholas wrote: Ok isn't this replacing a western hack with a bridge hack? The init 0 and init 6 probably aren't going to work anyway since (1) asterisk has to be running as root and I have already mentioned that this is a requirement. (2) the path in * is limited if even existent, so the init command would work unless you had a copy or symlink in the asterisk directory. # tr '\0' '\n' /proc/`cat /var/run/asterisk/asterisk.pid`/environ | grep ^PATH= PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin Init scripts tend to set the path explicitly. So those are just poor excuses for not using that fine hangup method. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/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] What is the purpose of membermacro in queues.conf
Rajkumar S wrote: Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj I'd be glad to explain them. First of all, setinterfacevar was actually around in 1.4, but its use has been expanded in 1.6.0. In 1.4, this would cause the MEMBERINTERFACE channel variable to be set. In 1.6.0, this setting also sets the MEMBERNAME, MEMBERCALLS, MEMBERLASTCALL, MEMBERPENALTY, MEMBERDYNAMIC, and MEMBERREALTIME variables. The purpose of exposing these values is to allow for an administrator to use these for any purpose he may desire. Second, there's setqueuevar. Its purpose is similar to setinterfacevar, in that it exposes values to the dialplan so that an administrator can use them how he wishes. The variables set are QUEUENAME, QUEUEMAX, QUEUESTRATEGY, QUEUECALLS, QUEUEHOLDTIME, QUEUECOMPLETED, QUEUEABANDONED, QUEUESRVLEVEL, and QUEUESRVLEVELPERF. Finally, you asked about membermacro. This allows for a macro to execute on a queue member's channel when he answers the call. This is very similar to the 'M' option for the dial application. Some people use this sort of feature as a post-answer hook into the dialplan so that they can perhaps log statistical information, or present the queue member with information about the incoming call. Mark Michelson ___ -- 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] Message
Title: Stylish Vacation You have told us you would like to receive exciting email offers from us. The sun is getting hotter, the days are getting longer, and summer vacations are almost here! Prepare to create some unforgettable memories with this stunning collection perfect to keep you Stylish On Vacation! Satisfaction Guaranteed - We want you to be absolutely satisfied with your pharmacy. Enjoy it. If, within 30 days after receipt of your purchase you're not completely satisfied, return it for the price you paid or we will gladly replace it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)
Some providers will give it to you on a PRI line. If you are using a TF number you'll get it regardless. However keep in mind that it takes me about 3 seconds to change outbound callerid. On Mon, Feb 16, 2009 at 9:10 PM, Alfred Monticello ajmce...@yahoo.com wrote: I'm thinking of starting a partyline, where people call in and talk to other people. For record keeping and billing purposes, I'd like to go by the callers telephone number. This method works fine as long as the caller doesn't have callerid blocked, but what are my options if they do block their number? I know there must be a way to report it, because there is a service provider here in my area that if I call and block my number, they are still able to obtain it. I know that when dialing a toll-free number, that the number is reported regardless. But what about regular non-toll free numbers? Does anybody have any ideas how I can do this? Are there any providers out there that offer this service over PRI or some other method? Thank you in advance. ___ -- 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] DTMF not completely muted
Michael Smith msmith at cbnco.com writes: Wilton Helm whelm at compuserve.com writes: There is no reason why it isn't possible to backup in the recorded message and erase the blip. Yes, that might be the way to go. I'm playing around with a modified __ast_play_and_record() that stops recording when the button is pressed, not released. I also have it hacking off the last 150ms of the recording if '#' is pressed. I'm not sure 150ms is enough, actually http://bugs.digium.com/view.php?id=14491 (now cutting the last 400ms) ___ -- 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] Questions about OpenSky - Asterisk to Skype Gateway
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote: Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp OpenSky is no related to any product from Digium. It is a different product. Today it's not as integrated as what is promised from Digium/ebay (it is is outbound only). However it is here today and it's free. It's also a hosted solution - meaning there's nothing to install or maintain on your Asterisk box. Just change a few lines in your configuration file (and even that is not necessary if you just dial a Skype name like this: nameofskypeu...@opensky.gizmo5.com). From: Alejandro Lengua alejandro.len...@gmail.com What about receiving Skype calls on Gizmo or other SIP device? Looking into the website I don't see anything regarding that. We are working on that capability and should have it shortly. Let me know if you want to be a tester. -- -- MR Michael Robertson www.MP3tunes.com - Your Music Everywhere www.Gizmo5.com - IM/VOIP/SMS from PC and phone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21.1 intermittent presence working with Polycom
Hi All, I upgraded a PBX from 1.2. to 1.4.21.1 and I'm noticing that the hints for SIP channels are not updating the phones 100% of the time. The hints seem to work for some time, then the notification on the phone will hang in either and on or off state. During this condition, on the PBX, core show hints, indicates the correct presence state for the SIP channel. Also if multiple phones are monitoring the same SIP channel, the presence notification on some phones still work fine, but may hang on one or two phones. We have to reboot the phone for the presence to start working again. We are using the same firmware on the phone that worked fine with the Asterisk 1.2 code, Polycom 650 with 2.1.1. So I'm guessing there is something particular with this version of Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Swift - detection of multiple digits unreliable on my system
Hi all,
I just installed Cepstral and app_swift version 1.4.2 on my Asterisk
1.4.22.1 box. It seems to work great with one exception.
If I play a test message with instructions to collect a maximum of 5
digits, it collects those 5 digits correctly if the user waits for the
message to complete before entering them. But if the user barges in
with digits before the message completes, the detected digits are
incorrectly (but consistently) detected. E.g., give the following AEL
context, if the user enters 60014 before the prompt completes,
saydigits says 612 every time.
context swiftTest {
s = {
answer;
wait(1);
swift(This is a test of Swift. Please enter your five
digit zip code.,1,5);
saydigits(${SWIFT_DTMF});
hangup;
};
};
Does this sound familiar to anyone? I am open to the possibility of
using swift -o to generate to a WAV file, then using that file with
read(), but I would like to avoid the delays and additional complexity
associated with that technique, if possible.
Thanks!
Bob Hartwig
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Re: [asterisk-users] Please help test the gender detection module at 575-613-4392
That's funny. The way I have it phrased, when I called I started talking to it as well! I have some code for short list voice recognition and thought about detecting yes and no in there, but I ran out of time...and the prompts were already recorded. Thank you everyone for helping test the module. There have been 200+ calls from users on the list and they are still coming in. We're getting about 65%-70% success rate. My target is 80%-85% in random sampling and 90%-95% in controlled settings. Update: I'm adjusting the detection ratios tomorrow, so that should improve general detection results based on the received data. I'm implementing filters to remove the background noise. I'd guess that 5% of those testing are trying to fool the system for fun, in one way or another. When the user is unaware of sampling, the results are slightly higher. My greeting suggests a less masculine phrase, but with a male voice. I suspect this throws off both genders' recordings. I probably should have had testers say their own names, since testers rarely divert on that. From: Gondar Monn gonda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 16, 2009 9:19:20 PM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 Looks like my provider is not passing dtmf correctly .. Had a serious laugh, system kept asking me if I was ready., ended up finding myself talking to the IVR . ___ -- 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] Credit Card processing machines
The ADT alarm going thru VoIP will create a life safety issue. Hope you planned for that.. --Don On 2/17/09 6:31 AM, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 17 Feb 2009, Andrew Joakimsen wrote: On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote: Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look into reducing the baud rate on the machines, but sadly the bank has them password protected and wants to charge a reprogramming fee :( They make credit card terminals with Ethernet -- use that instead. The client's processor charges 7c/transaction over IP (plus normal charges), so they are quite keen to keep it working the way it was before I replaced their PBX ;) As a followup, *99 prepended on any Linksys ATA does indeed make a difference in modem reliability. Both their CCs and their ADT alarm devices now function reliably. I also reduced the CC baud rate to 300 baud (!), and it is rock solid now! j ___ -- 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] Question regarding custom announcements in queues.conf
Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? 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] Stress Testing IVR
Rajkumar S schrieb: How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be programmed to send out digits after specified time to simulate users pressing menu items. You could remotely control a Snom 3xx like that. But I guess that's not what you're looking for. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? SIPp: http://sipp.sourceforge.net/ Any other method to stress test an IVR call flow? Call, call, call ... ;-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Question regarding custom announcements in queues.conf
Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson ___ -- 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] Lost with Patton 5.3 web server. Registration ?
Hi, How do you configure a Patton smartnode to register with an Asterisk server ? I could do it with 4.2 web server but I'm lots with 5.3 web interface ? Alternatively, has anyone a correct running-config for that ? Regards ___ -- 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] Question regarding custom announcements in queues.conf
Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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 supports SIP-T?
Asterisk supports SIP-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] Packet Truncated - Choppy Audio
Hello Danny, Thank you for the swift reply! As it turns out, this was an artifact from ntop, which has a default maximum buffer size of 8232 bytes. We're still getting choppy audio, but we've ruled this error message out as a possible cause. Thanks again, Matt. From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Packet Truncated - Choppy Audio This indicates that your NIC card is not handling the throughput effectively. Is * the only application on your server? How many users are on * when this occurs? ___ -- 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] Question regarding custom announcements in queues.conf
Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) On Tue, Feb 17, 2009 at 1:05 PM, Mark Michelson mmichel...@digium.comwrote: Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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] Network architecture
found out that the best solution is to use OpenSips as SIP OpenSIPS is a great free software proxy. 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? There isn't any explicit limitation in Asterisk or OpenSIPS that I'm aware of, but you are limited to processing power, memory, bandwidth, etc. 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? There are a lot of factors to consider, but I'm sure you could do it if you are determined. Not the wisest option however - see below. 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? I would split it up and keep each server under 50% load during normal activity. That way you can handle peak load and balance if one or more servers fail. Try not to put more than 200-400 calls on each server, depending on your configuration. That would be 100-200 calls per server with 50% load. For 5,000 concurrent calls, that means 25 servers assuming decent hardware and 50% load. That might not be an option. You may be able to split up some of the servers into multiple VMs -- maybe five servers with five VMs each. You may be able to get away with 90% regular load if 5,000 concurrent calls is never to be exceeded. Anyway, there are many factors to consider. More information is definitely needed. From: michel freiha mich...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; asterisk-users-boun...@lists.digium.com Sent: Tuesday, February 17, 2009 7:19:58 AM Subject: [asterisk-users] Network architecture Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other stuff plus a MySQL database... This architecture convinced me, but I have some questions regarding asterisk and I need asterisk expert answers in order to take decision... 1- Is there any Software limitation on asterisk regarding number of simulltaneous calls? 2- Can 1 asterisk server handle 5000 simuitaneous calls if I have the appropriate hardware? 3- It's etter to have one asterisk server for hadling 5k simultaneous calls or divide the load on different servers? Waiting your reply Regards ___ -- 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] Credit Card processing machines
Certainly a sobering thought. Have others had to deal with this in PBX replacement scenarios? Its a giant cost savings in this case - they are dropping about 12 POTS lines in favor of utilizing (an underutilized) T1 trunk that was already in place. j On Tue, 17 Feb 2009, Don E. Wisdom wrote: The ADT alarm going thru VoIP will create a life safety issue. Hope you planned for that.. --Don On 2/17/09 6:31 AM, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 17 Feb 2009, Andrew Joakimsen wrote: On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote: Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look into reducing the baud rate on the machines, but sadly the bank has them password protected and wants to charge a reprogramming fee :( They make credit card terminals with Ethernet -- use that instead. The client's processor charges 7c/transaction over IP (plus normal charges), so they are quite keen to keep it working the way it was before I replaced their PBX ;) As a followup, *99 prepended on any Linksys ATA does indeed make a difference in modem reliability. Both their CCs and their ADT alarm devices now function reliably. I also reduced the CC baud rate to 300 baud (!), and it is rock solid now! j ___ -- 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] Question regarding custom announcements in queues.conf
Christopher Aloi wrote: Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) Yes, if an absolute path is not provided for the sounds, then it is assumed that the default sounds directory is where the sound may be found. I just tried a small test on that revision of 1.4, and it worked for me. In my case, I was simply playing the beep sound file which already exists in the sounds directory. Mark Michelson ___ -- 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] Question regarding custom announcements in queues.conf
Yah - Found my problem, I can't spell - periodic-*annouce* = SD-PLS-HOLD periodic-announce-frequency=10 : ) On Tue, Feb 17, 2009 at 1:19 PM, Christopher Aloi chris.a...@gmail.comwrote: Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) On Tue, Feb 17, 2009 at 1:05 PM, Mark Michelson mmichel...@digium.comwrote: Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto: 7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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] Question regarding custom announcements in queues.conf
Christopher Aloi wrote: Yah - Found my problem, I can't spell - periodic-*annouce* = SD-PLS-HOLD periodic-announce-frequency=10 : ) Oh, Ha! That'll do it every time. Mark Michelson ___ -- 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] Credit Card processing machines
snip Certainly a sobering thought. Have others had to deal with this in PBX replacement scenarios? Its a giant cost savings in this case - they are dropping about 12 POTS lines in favor of utilizing (an underutilized) T1 trunk that was already in place. /snip Yes -- our alarm monitoring company considers T1 - * - ATA - Alarm to be so unreliable that they require you to sign a waiver (indemnifying them in the event of basically anything) if you hook it up this way. Because of that we kept a POTS line around to hook up the alarm system. It would be cheaper to hook the alarm panel up to an internal cell phone backup :). I assume there are manufacturers that offer a built-in cell modem... --Dave ___ -- 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] Question regarding custom announcements inqueues.conf
Can live in this directory or any under it. If you specify file * looks in VLAS, if you specify foo/file * looks in VLAS/foo. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Aloi Sent: Tuesday, February 17, 2009 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question regarding custom announcements inqueues.conf Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) On Tue, Feb 17, 2009 at 1:05 PM, Mark Michelson mmichel...@digium.com wrote: Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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] Please help test the gender detection module at 575-613-4392
For those who testing the gender detection module via the number provided: How was the experience, aside from the funny beep? In your perception, how well did it perform? (I see raw numbers here, but perception is important too.) Do you have any comments, suggestions, or feedback? From: Asterisk Asterisk nt_aster...@yahoo.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Gondar Monn gonda...@gmail.com; nt_aster...@yahoo.com; nt_jnew...@yahoo.com Sent: Tuesday, February 17, 2009 9:10:38 AM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 That's funny. The way I have it phrased, when I called I started talking to it as well! I have some code for short list voice recognition and thought about detecting yes and no in there, but I ran out of time...and the prompts were already recorded. Thank you everyone for helping test the module. There have been 200+ calls from users on the list and they are still coming in. We're getting about 65%-70% success rate. My target is 80%-85% in random sampling and 90%-95% in controlled settings. Update: I'm adjusting the detection ratios tomorrow, so that should improve general detection results based on the received data. I'm implementing filters to remove the background noise. I'd guess that 5% of those testing are trying to fool the system for fun, in one way or another. When the user is unaware of sampling, the results are slightly higher. My greeting suggests a less masculine phrase, but with a male voice. I suspect this throws off both genders' recordings. I probably should have had testers say their own names, since testers rarely divert on that. From: Gondar Monn gonda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 16, 2009 9:19:20 PM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 Looks like my provider is not passing dtmf correctly .. Had a serious laugh, system kept asking me if I was ready., ended up finding myself talking to the IVR . ___ -- 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] Please help test the gender detection moduleat 575-613-4392
After helping out it seems I've been determined a female(wrongly). It was disappointing and I'm considering a visit to the Dr Phil Show to work out my anger From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Asterisk Sent: Tuesday, February 17, 2009 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: nt_jnew...@yahoo.com Subject: Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392 That's funny. The way I have it phrased, when I called I started talking to it as well! I have some code for short list voice recognition and thought about detecting yes and no in there, but I ran out of time...and the prompts were already recorded. Thank you everyone for helping test the module. There have been 200+ calls from users on the list and they are still coming in. We're getting about 65%-70% success rate. My target is 80%-85% in random sampling and 90%-95% in controlled settings. Update: I'm adjusting the detection ratios tomorrow, so that should improve general detection results based on the received data. I'm implementing filters to remove the background noise. I'd guess that 5% of those testing are trying to fool the system for fun, in one way or another. When the user is unaware of sampling, the results are slightly higher. My greeting suggests a less masculine phrase, but with a male voice. I suspect this throws off both genders' recordings. I probably should have had testers say their own names, since testers rarely divert on that. From: Gondar Monn gonda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 16, 2009 9:19:20 PM Subject: Re: [asterisk-users] Please help test the gender detection module at 575-613-4392 Looks like my provider is not passing dtmf correctly .. Had a serious laugh, system kept asking me if I was ready., ended up finding myself talking to the IVR . - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] Credit Card processing machines
David Gibbons wrote: snip Certainly a sobering thought. Have others had to deal with this in PBX replacement scenarios? Its a giant cost savings in this case - they are dropping about 12 POTS lines in favor of utilizing (an underutilized) T1 trunk that was already in place. /snip Yes -- our alarm monitoring company considers T1 - * - ATA - Alarm to be so unreliable that they require you to sign a waiver (indemnifying them in the event of basically anything) if you hook it up this way. Because of that we kept a POTS line around to hook up the alarm system. It would be cheaper to hook the alarm panel up to an internal cell phone backup :). I assume there are manufacturers that offer a built-in cell modem... lots of that cell modem stuff, but the latest trend is to have constant connectivity over the internet instead of a dedicated serial link over something like DVACS that can detect line cuts. A normal alarm is only connected when it has something to report unless its a higher end system connected all the time. on the credit card terminals internet connectivity is also becoming standard since many units can all share and don't need an aggregator or dedicated phone lines. --Dave ___ -- 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] Credit Card processing machines
On Tue, 17 Feb 2009, Jon Pounder wrote: Yes -- our alarm monitoring company considers T1 - * - ATA - Alarm to be so unreliable that they require you to sign a waiver (indemnifying them in the event of basically anything) if you hook it up this way. Because of that we kept a POTS line around to hook up the alarm system. It would be cheaper to hook the alarm panel up to an internal cell phone backup :). I assume there are manufacturers that offer a built-in cell modem... lots of that cell modem stuff, but the latest trend is to have constant connectivity over the internet instead of a dedicated serial link over something like DVACS that can detect line cuts. A normal alarm is only connected when it has something to report unless its a higher end system connected all the time. on the credit card terminals internet connectivity is also becoming standard since many units can all share and don't need an aggregator or dedicated phone lines. That is in fact the way they went for the remote stores, as we couldn't make it work reliably over the net back to their main office (this is in the Virgin Islands, where connectivity is expensive, slow, and unreliable at best). But processors down there make you pay dearly for the right to do so. We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! j ___ -- 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] Credit Card processing machines
snip We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! /snip We do *fax* in this way and it works like a charm. We can hit much more than 2400 baud I think too. --Dave ___ -- 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] Network architecture
You may be able to split up some of the servers into multiple VMs -- maybe five servers with five VMs each. I'm not sure I see the merit in this. VMs seem to be regarded as a magic bullet (i.e. free lunch). I don't know of any case where 5 VMs can accomplish more work on one processor than simply letting the processor manage it all (except if the OS and or application can't efficiently split the task into the necessary multiple threads, which I don't think is an issue here). By definition, the total accomplished must be less with VMs, because the hypervisor will take some CPU cycles. Wilton ___ -- 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] Credit Card processing machines
On Feb 17, 2009, at 1:20 PM, David Gibbons wrote: snip We will be testing the ADT connection heavily this week. The modem connections to my understanding are 2400 baud. Over G.711U and a T1 I don't see why this wouldn't be as solid as a POTS line, but our tests will tell! /snip We do *fax* in this way and it works like a charm. We can hit much more than 2400 baud I think too. --Dave Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. ___ -- 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] Credit Card processing machines
On Tue, 17 Feb 2009, Jerry Jones wrote: Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. You mean they communicate with an IVR? Seems like that could be made solid with the right DTMF options enabled on the ATA. FWIW that makes a lot more sense than a modem connection. j ___ -- 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] call file bug?
I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as Channel: DAHDI/1/8775203463 When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to DAHDI/1/8775203463 it'll success. . If I change the channel to SIP/8000 and put the call file under /var/spool/asterisk/outgoing it is also success - it calls the extension 8000 and the controle goes to the context after the extension 8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this release? Thanks. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
On Tue, Feb 17, 2009 at 15:09, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 17 Feb 2009, Jerry Jones wrote: Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. You mean they communicate with an IVR? Seems like that could be made solid with the right DTMF options enabled on the ATA. FWIW that makes a lot more sense than a modem connection. No, it's not an IVR. It's a protocol called ContactID. ___ -- 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] Credit Card processing machines
Jeff LaCoursiere wrote: On Tue, 17 Feb 2009, Jerry Jones wrote: Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. You mean they communicate with an IVR? Seems like that could be made solid with the right DTMF options enabled on the ATA. FWIW that makes a lot more sense than a modem connection. j ___ -- 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 If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. Jonn ___ -- 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] Credit Card processing machines
On Tue, 17 Feb 2009, Andrew Joakimsen wrote: Most alarm systems around here use bursts of dtmf - not an actual modem to communicate with alarm central. Yes I have seen these have many issues with voip in the path. You mean they communicate with an IVR? Seems like that could be made solid with the right DTMF options enabled on the ATA. FWIW that makes a lot more sense than a modem connection. No, it's not an IVR. It's a protocol called ContactID. Ahh. I just read a PDF on the protocol. It may as well be an IVR - it is all standard DTMF with normal DTMF timing between digits. Where does VoIP introduce a problem? Seems like this should work well. j ___ -- 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] Credit Card processing machines
On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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] call file bug?
You should post the call file. Also, I'd use DAHDI/G1 instead of DAHDI/1 as that ties the call to a specific port/line (perhaps what you want to do?) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ray Chen Sent: Tuesday, February 17, 2009 2:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] call file bug? I have a problem of using call file to make an auto dial out call through FXO channel. I defined the channel in the call file as Channel: DAHDI/1/8775203463 When I put the call file under the /var/spool/asterisk/outgoing dir it did not call out but came to the context I defined in extensions.conf as if the callee had answered the call. If I make a call from an extension to DAHDI/1/8775203463 it'll success. . If I change the channel to SIP/8000 and put the call file under /var/spool/asterisk/outgoing it is also success - it calls the extension 8000 and the controle goes to the context after the extension 8000 answers the call. I am using asterisk 1.4.23.1. Is it a bug introduced in this release? Thanks. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com http://www.mail.com/Product.aspx ! ___ -- 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] Please help test the gender detection moduleat 575-613-4392
Accuracy should be 10%-15% better on Wed or Thu. From: Jason Aarons (US) jason.aar...@us.didata.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 17, 2009 10:48:07 AM Subject: Re: [asterisk-users] Please help test the gender detection moduleat 575-613-4392 After helping out it seems I’ve been determined a female(wrongly). It was disappointing and I’m considering a visit to the Dr Phil Show to work out my anger…. ___ -- 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] only ring phones that are not on a call
is there anything i can do in my dialplan to only ring phones which are not on a call at the time someone dials in? its for a call center, they do not want to use queues, but they are complaining that the call waiting beep is annoying. i tried call-limit in the sip.conf but then it just busy out all phones when a call comes in. any thoughts? thanks, jon ___ -- 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] Credit Card processing machines
National fire protection association They write the fire codes. http://www.nfpa.org On 2/17/09 1:28 PM, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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] Updated modules to be released (FaxDetect, GenderDetect, MachineDetect, others)
I will be releasing updated versions to many of the detection modules next week. They include better support of Asterisk 1.2, 1.4, and 1.6, better detection, better parameters, an easier build system, and usability is enhanced. The updated modules include: * FaxDetect, LineDetect, and MachineDetect - which many are presently using * PlayDetect and BackgroundDetect - playback with specification of detection modules to use * GenderDetect, NoiseDetect, and AnswerDetect - new modules Contact me off the list if you need updated modules or have questions, comments, or feedback. Justin Newman nt_jnewman at yahoo.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Credit Card processing machines
Jeff LaCoursiere wrote: On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? national fire protection association. and the internet connection is one way to solve that since it acts like a dedicated line with constant yes everything is ok packets, not just communication during a problem. j ___ -- 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] life safety system and VOIP
http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] life safety system and VOIP
What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] life safety system and VOIP
Jason Aarons (US) wrote: In general in the terminology for this stuff supervised just means the system its referring to not only knows when something bad is happening, it also is constantly told everything is ok, and timing out waiting for that ok is also an indication of a problem. There is nothing magic about it and there are many different ways it can be accomplished on various media that all satisfy the regulations, but in general a dialup on demand connection whether it be voip or copper does not satisfy supervised as a requirement. Contact ID is just one protocol, fsk is another that is basically modemlike. Most alarms can be configured for a handful of protocols. There is even a channel for asterisk made for receiving one of them (I forget which) and I think its nextalarm.com that is using it for their monitoring with voip boxes. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] Caller Hangup detection
Hello, Is here any dial plan variable which could help me to identify that call was dropped (when still not connected) by caller? HANGUPCAUSE returns 0 DIALSTATUS returns NOANSWER How to identify such situation? Related question - how to know which end (caller or callee) ended the call first after call was answered? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] only ring phones that are not on a call
On Tue, 17 Feb 2009, Jon Weisman wrote: is there anything i can do in my dialplan to only ring phones which are not on a call at the time someone dials in? its for a call center, they do not want to use queues, but they are complaining that the call waiting beep is annoying. i tried call-limit in the sip.conf but then it just busy out all phones when a call comes in. any thoughts? Can't you simply turn call-waiting off on the phones? 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] life safety system and VOIP
Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] life safety system and VOIP
I think the BAT SIGNAL is the answer. POTS lines have their issues as well - how many times did we redial to get into our ISP's in the mid nineties? I have trouble believing the fire code actually spells out that dedicated POTS lines must be used. Regradless I think another hold harmless just made it into my service contract. j On Tue, 17 Feb 2009, Jon Pounder wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] life safety system and VOIP
On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co's ATM network or the cable companies network (or lack there of) --Don j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] only ring phones that are not on a call
its about 400 phones, and i dont have access to the tftp server. i was just looking for a faster way. thanks, jon - Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 17, 2009 4:05 PM Subject: Re: [asterisk-users] only ring phones that are not on a call On Tue, 17 Feb 2009, Jon Weisman wrote: is there anything i can do in my dialplan to only ring phones which are not on a call at the time someone dials in? its for a call center, they do not want to use queues, but they are complaining that the call waiting beep is annoying. i tried call-limit in the sip.conf but then it just busy out all phones when a call comes in. any thoughts? Can't you simply turn call-waiting off on the phones? 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 ___ -- 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] life safety system and VOIP
On Tue, 17 Feb 2009, Don E. Wisdom wrote: In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co's ATM network or the cable companies network (or lack there of) This depends heavily on where you are. In the Virgin Islands the most reliable Internet access is served wireless, with dedicated radios on the roof. Everyone has a diesel generator because the power goes out all the time. The phone company (that happens to be digging itself out of chapter 11 right now) has just as bad a reputation, and the last time there was a bad hurricane, the only service that was working was the Internet link. Of course when your diesel runs out you are SOL... j ___ -- 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] life safety system and VOIP
Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co’s ATM network or the cable companies “network” (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for
Re: [asterisk-users] life safety system and VOIP
Jeff LaCoursiere wrote: I think the BAT SIGNAL is the answer. POTS lines have their issues as well - how many times did we redial to get into our ISP's in the mid nineties? I have trouble believing the fire code actually spells out that dedicated POTS lines must be used. its the supervised that is relevent, not dedicated,. its not used as a dialup line, basically its just connected period and if it goes away for whatever reason the monitoring station knows immediately. In Canada the technology is called DVACS and its basically just modems on the ends of a dry copper pair, not sure what its called elsewhere. I think at the monitoring station the dry pairs are not really dry but aggregated into some Supermodem kind of like the t1 equivalent of pots lines, I don't really know though. The technology is definately on the way out, and being replaced with the tcp based stuff to accomplish the same thing. Regradless I think another hold harmless just made it into my service contract. j On Tue, 17 Feb 2009, Jon Pounder wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] only ring phones that are not on a call
You could set up a hint for the extension and check the hint for inuse
before executing the Dial in your dialplan
Exten = 801,hint,SIP/100
Exten = XXX,1,System(/usr/sbin/asterisk -rx core show hints|/bin/grep
SIP/100|/bin/grep InUse
Exten = XXX,2,GOTOIF($[{CMSTATUS} = FAILURE])?dial
Exten = XXX,3,hangup()
Exten = XXX,4(dial),Dial(SIP/100)
Exten = XXX,5,hangup()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Weisman
Sent: Tuesday, February 17, 2009 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] only ring phones that are not on a call
is there anything i can do in my dialplan to only ring phones which are not
on a call at the time someone dials in?
its for a call center, they do not want to use queues, but they are
complaining that the call waiting beep is annoying.
i tried call-limit in the sip.conf but then it just busy out all phones when
a call comes in.
any thoughts?
thanks,
jon
___
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[asterisk-users] ztdummy compile under 2.6.28 ?
It looks like something has changed in the HPET kernel code in 2.6.28 (maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4 versions of zapata) A kernel structure member has been renamed with some crypic comments in the lkml about it. Anyone know the right thing to do - I'm not up on the linux kernel guts, nor how ztdummy might interact with it, so simply renaming the structure member (from expires to _expires) is probably not the right thing to do... Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC instead? 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
[asterisk-users] SLA and Flashing BLF
I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the MeetMe? The current behaviour is to leave the line BLF solid, not flashing. -- Muiz Motani m...@askaritech.com Askari Technologies ___ -- 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] life safety system and VOIP
Jon Pounder wrote: Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co’s ATM network or the cable companies “network” (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. This is true, that is why most fire panels have to have 2 phone lines. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section. 8.5.2.2* Alternate Methods. 8.5.4 Other Transmission Technologies. 8.6.2.2* Alternate Methods. 8.6.4 Other Transmission Technologies. There is nothing specific with regards to voice over internet protocal and leaves room to add new technology proposals with requirements in future editions according to A8.5.2.2. or A8.6.2.2 respectively. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, February 17, 2009 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Credit Card processing machines On Tue, 17 Feb 2009, Jonn Taylor wrote: If you are in the US, ANY life safety system has to be connected to a dedicated copper POTS line. VOIP is NOT ok to use for this. It is in the NFPA. What is the NFPA? Do analog extensions in traditional PBXes count? j ___ -- 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] ztdummy compile under 2.6.28 ?
On Tue, Feb 17, 2009 at 5:00 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC instead? The best way to use ztdummy is to read about the change to using DAHDI, and use dahdi_dummy instead. http://www.voip-info.org/wiki/view/DAHDI ___ -- 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] ztdummy compile under 2.6.28 ?
Gordon Henderson wrote: Anyone know the right thing to do - I'm not up on the linux kernel guts, nor how ztdummy might interact with it, so simply renaming the structure member (from expires to _expires) is probably not the right thing to do... If you're already making system changes and updating your kernel, now might be a good time to make the switch to dahdi. dahdi_dummy works with recent kernels. But otherwise, you might be interested in the history of dahdi_dummy.c to get some hints about what you might need to do for ztdummy if you want to make a local patch for that. http://svn.digium.com/view/dahdi/linux/trunk/drivers/dahdi/dahdi_dummy.c?view=log Cheers, Shaun ___ -- 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] Stress Testing IVR
On Tue, Feb 17, 2009 at 1:51 AM, Rajkumar S rajkum...@gmail.com wrote: How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be programmed to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Asterisk is your friend at generating large numbers of simultaneous calls. Read up on call files and bash loops. As for actually putting delays and pressing the right buttons, you're on your own. You would need to write a custom AGI script specific to your IVR, and call it from your call file, which you then put in a bash loop. In that case, DTMF is your friend. Does any one have any recommendations ? Any other method to stress test an IVR call flow? I swear I've heard of a softphone that would listen to your actions and then you could replay them, but I'm not sure whether I dreamed that up or it really exists. ___ -- 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] life safety system and VOIP
In Florida some new subdivision developers have sold the phone/cable/internet rights to a provider. They run fiber to each house and then have the uplink to provider which isn't a traditional telco. You can't get another provider as satellite dishes are limited in covenants and restrictions (CCR). I guess you could get GSM or CDMA service from cell provider or WiMax/LTE. It provides an upfront funding to developer for sewer/water costs. I'd be curios what battery life they have. I know the FCC mandated cell towers have more battery life after Hurricane Katrina wiped out communications in New Orleans for months. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonn Taylor Sent: Tuesday, February 17, 2009 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] life safety system and VOIP Jon Pounder wrote: Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net mailto:j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co's ATM network or the cable companies network (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. This is true, that is why most fire panels have to have 2 phone lines. j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter
Re: [asterisk-users] ztdummy compile under 2.6.28 ?
On Tue, 17 Feb 2009, David Backeberg wrote: On Tue, Feb 17, 2009 at 5:00 PM, Gordon Henderson gordon+aster...@drogon.net wrote: Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC instead? The best way to use ztdummy is to read about the change to using DAHDI, and use dahdi_dummy instead. http://www.voip-info.org/wiki/view/DAHDI Thanks, but I'm sticking to 1.2 for the time being. I might look at what's changed in dhadi though, and I'll switch to it when I can type it without making a typo. 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] life safety system and VOIP
Jonn Taylor wrote: Jon Pounder wrote: Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co’s ATM network or the cable companies “network” (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. Obviously you are unaware of the very many SLIC cabinets and vaults in use in the US. Fewer and fewer dial tone comes directly from the CO. He is correct. These are remote D to A converters that are at the mercy of the batteries in the remotes, some last 4 hours, if they are maintained. In other areas the Telco's have to scramble with portable generators to keep service up. In other cases even the CO's can't outlast the devastation of an ice storm, and have to have power brought in, all assuming the local Telco is able to. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. This is true, that is why most fire panels have to have 2 phone lines. True, but when both lines are served from the same CO, over the same cable, it is really a false sense of security. In the US also, dry copper supervised pairs are scarce as hens teeth any more. Time was a copper pair was supervised with a DC current from end to end, and if something would open the circuit, that alerted the monitoring station there was a trouble. If there was a real alarm, they DC was reversed, and the monitoring station would react accordingly. Ancient history now. Dry pairs have disappeared over the last 20-30 years, and many other schemes have come and gone. Few UL and NFPA systems allow VOIP though. Risk management still considers it unreliable, and of course, they are correct. Anyone who believes otherwise, ask your business insurance provider for a ruling. John Novack j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper to all the motorist aid boxes along the highway. I thought most of your alarm panels have moved to GSM/CDMA backup communications. I'd like to see a fire marshall not give a permit for having a VoIP ATA or Vonage. http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 It's permitted in Chapter 8 2002 2007 Alternative Methods of Communication and these still have supervision in accordance with Chap 4 and it's sub-section.
Re: [asterisk-users] SLA and Flashing BLF
Muiz Motani wrote: I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the MeetMe? The current behaviour is to leave the line BLF solid, not flashing. Actually, the code does set the device state of the MeetMe to 'hold', but not all SIP phones can display a 'hold' state on a line key using the method we use for signaling to them. What brand of phones are you using, and are you using up-to-date firmware for them? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phone 7940G.
Here's an alternative to TFTP that works with Polycom 501's. Enable HTTP in *. Under your static-http directory make a phones dir and put your files there. In the phone setup, select HTTP and point to http://1.2.3.4:8088/asterisk/static-http/phones changing 1.2.3.4 to your local * IP. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ronny Julian Sent: Friday, February 13, 2009 8:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phone 7940G. Please send on! Thanks! What TFTP server did you use? Catalin S. wrote: hey finally i did it. I upgraded the firmware to the latest sip firmware and now i have the another problem. The requested files are the following: ---///--- Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf to 192.168.1.3:51253 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to 192.168.1.3:51254 ---///--- I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is the mac address of phone, but i don't know what to write in CTLSEP00141CAA4B4C.tlv, SEP00141CAA4B4C.cnf.xml and SIPDefault.cnf. On display of the screen of phone all I have is Tftp file missing... probably it expect all these files. Anyway, Ronny I can give you my archive with what i had in my tftp and i succeeded to update firmware. Just tell me if you want to send on your personal e-mail these files. Thank you guys for your help and interest. On 2/13/09, k4...@bellsouth.net mailto:k4...@bellsouth.net k4...@bellsouth.net wrote: I'm trying to do the same and have read the mentioned sites. The one item I can't seem to get past is a working TFTP server. What is the easiest method to get one running and what packages in Linux or Windows work best? Thanks for putting up with a Linux newbie. Ronny -- Original message from Alex Balashov mailto:abalas...@evaristesys.com abalas...@evaristesys.com: -- Have a look at: http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a 0080 094584.shtml#topic2 On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S. wrote: I understand, but i cannot load the new firmware... is any well know method? On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov wrote: This phone is currently running the SCCP (Skinny) image. Before you will get anywhere you need to load the SIP firmware image onto it. The SEP* configuration files are for SCCP. After doing that, the phone will start requesting the correct files. You may need to upgrade through various SIP images cumulatively. On Fri, 13 Feb 2009 11:42:03 +0200, Catalin S. wrote: Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get any luck here either. I look in the /var/log/messages and I observed that my phone request 4 different files that i don't have it in my tftp directory. Here's my tftp output session with my phone: Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving OS79XX.TXT to 192.168.1.3:52178 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:52179 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving XMLDefault.cnf.xml to 192.168.1.3:52180 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf to 192.168.1.3:52181 Feb 13 11:19:36 linux-9pg5 atftpd[18825]: Serving SEPDefault.cnf to 192.168.1.3:52182 as you see my phone request 4 files that doesn't comes in archive P0S3-08-11-00.zip: SEP00141CAA4B4C.cnf.xml, XMLDefault.cnf.xml, SEP00141CAA4B4C.cnf, SEPDefault.cnf... while my archive contents is the following: OS79XX.TXT, P003-08-11-00.bin, P003-08-11-00.sbn, P0S3-08-11-00.loads, P0S3-08-11-00.sb2 3.) I want to make this phone to be SIP compatible. A friend of main gave me a .cnf file with an example of configuration for SIP. How may I rename this cnf file to make work with my phone. 4.) On the other side my phone doesn't have ringtone either. Any clue how may I put ringtones on it? I know is a lot of questions for you guys, but I browse on cisco.com web site and google for hours and I don't get it any clue to make work this phone in any way. Thank you for help. Jonson. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] life safety system and VOIP
John Novack wrote: Jonn Taylor wrote: Jon Pounder wrote: Don E. Wisdom wrote: On 2/17/09 2:05 PM, Jon Pounder j...@inline.net wrote: Jeff LaCoursiere wrote: What do you suppose we have as liability if we are asked to install such systems? Is it the responsibility of the business owner that orders the system to meet all applicable codes? If (god forbid) someone was hurt in such a situation and the alarm didn't get passed because of being delivered by VoIP for whatever reason, does the system installer have any liability? well here's a question - which is more reliable ? - a single copper line dialed on demand when there is a problem - voip or other internet technology, using internet connections on more than one media (say phone and cable), voip connected to multiple servers in a failover configuration. its not uncommon for even a house to have multiple internet connections, but how many buildings have phone lines that connect back to different CO's and fail over ? The best bet if you really care about what you are trying to protect is make sure the message can get out as many ways as possible, whether it be phone, voip, network, cellmodem, etc. Forget what regulations require, no one says you can't go further than the minimum if you want. In a REAL emergency internet/cell is more likely to fail than the phone companys pots network. Cable/DSLAM etc only have about 4 hours of battery power. The CO has a entire battery room which will last a whole lot longer. Not to mention that it may stay up longer than your VoIP network. You also have to take into account everything between you the CO or cable company. If just ONE thing fails you loose voip. Copper is a lot more forgiving has failover modes versus the phone co’s ATM network or the cable companies “network” (or lack there of) --Don I don't know if thats really true any more, all the new areas around here have satellite CO's where fibre comes out to a box on the street with some batteries etc and copper runs out from there - great for dsl since its close, but at the mercy of whatever batteries are in there. The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. Obviously you are unaware of the very many SLIC cabinets and vaults in use in the US. Fewer and fewer dial tone comes directly from the CO. He is correct. These are remote D to A converters that are at the mercy of the batteries in the remotes, some last 4 hours, if they are maintained. In other areas the Telco's have to scramble with portable generators to keep service up. In other cases even the CO's can't outlast the devastation of an ice storm, and have to have power brought in, all assuming the local Telco is able to. I am very aware of how the public telephone network works as our company installs CO's for many different telephone companies all over the US. Yes some of them install all of the equipment in the remote cabinets and others do not. Some do fiber to home. They all have batteries that can fail. maybe your alarm needs to report in since there is a fire in your phone equipment - what then ? I have seen every type of media go down or have problems no matter how stable - the only answer is have more than one so you always have a backup. Poles get hit, cables get cut, equipment breaks, its just a fact of life. This is true, that is why most fire panels have to have 2 phone lines. True, but when both lines are served from the same CO, over the same cable, it is really a false sense of security. In the US also, dry copper supervised pairs are scarce as hens teeth any more. Time was a copper pair was supervised with a DC current from end to end, and if something would open the circuit, that alerted the monitoring station there was a trouble. If there was a real alarm, they DC was reversed, and the monitoring station would react accordingly. Ancient history now. Dry pairs have disappeared over the last 20-30 years, and many other schemes have come and gone. Not true!!! The telephone companies today are driven by money. They still can provide dry pairs. They just do not want to, its not in their best interest. Few UL and NFPA systems allow VOIP though. Risk management still considers it unreliable, and of course, they are correct. Anyone who believes otherwise, ask your business insurance provider for a ruling. This is very true. Anyone ever read the disclaimer from vonage? John Novack j On Tue, 17 Feb 2009, Jason Aarons (US) wrote: http://www.iccsafe.org/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=1;t=001650 ;p=1 I can't see the Dept Transportation running copper
Re: [asterisk-users] life safety system and VOIP
The dial tone for the phone line still comes from the CO. The phone companies loop there copper cable in and out of the remote cabinets. Remote terminals are served by T1 or higher density carrier circuits, which can be either copper or fiber, often employing statistical multiplexing. While the DT may originate in the CO, it does so only in a data sense, not an analog POTS sense. The remote terminal actually generates the POTS analog signal, and is dependent on the life of the batteries in the box. They are good for several hours, maybe even a day, but definitely not weeks. Some RTs also have a DSLAM associated with them for DSL, but that is a separate topic and involves more batteries. This is true, that is why most fire panels have to have 2 phone lines. Which only catches about half of the problems, assuming both come through the same cable from the same CO or RT (and, in the latter case, the same carrier circuit). If a card fails or the I R guy opens or shorts the loop, the other line can take over. If the CO or RT crashes, or batteries die or cable gets dug through by a backhoe, guess what goes down! For serious mission critical circuits the engineer specifies two different operating companies and requires each to provide complete circuit details so he can insure that one isn't leasing lines from the other, or other scenarios that would be vulnerable to a single incident. Time was a copper pair was supervised with a DC current from end to end, Another variation on this theme used by central alarm monitoring companies of years ago was to have the telco provide a copper loop that included a number of customer sites. Basically each site was in series. At the monitoring station was the DC power and a relay. If all was well the loop was complete and the relay operated. Each site had a mechanical interrupter--a spring wound gear mechanism that pulsed out digits by breaking the loop momentarily. When an alarm condition occurred (such as water movement in a sprinkler riser) the spring would wind down, turning the gears and pulsing opens on the loop. In some cases, this caused ink mark square waves that could be counted on paper. The pulses were similar to rotary dial pulses in groups for digits, but slower speed. They represented the ID number of the sender reporting, which identified the customer and location. Of course, if anything in the loop, any sender, any telco drop, failed, the whole set of customers was unmonitored until it was fixed--which could be a day or two in extreme cases. I was called out once to service a site that had these. The one good thing about them was the only electrical requirement was at the monitoring station. Wilton ___ -- 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 supports SIP-T?
On Tue, Feb 17, 2009 at 1:06 PM, Daviramos Roussenq Fortunato daviramo...@gmail.com wrote: Asterisk supports SIP-T? Nope. Here is some old discussion on this topic: http://lists.digium.com/pipermail/asterisk-biz/2008-May/026690.html -- Raj Jain ___ -- 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] Open Source in an Economic Downturn: Asterisk stories needed
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on Sunday in Los Angeles, and the topic of my talk is Open Source in an Economic Downturn. I've got lots of talking points for this talk, but it would be interesting to hear some short anecdotes about how you in the Asterisk community are thriving, or at least surviving, by virtue of the benefits of Open Source. I find that real-world examples are worth more than all of the bullet points in the world, and timely stories from the community would be more interesting than hearing me prattle on. Please ensure that your snippet or list of points are in some way related to the benefits of open-source, or how other alternatives are less attractive in the cough compressed economic environment. I'd prefer of course to hear about how Asterisk is the silver bullet for your particular business, but I'm open to any OSS-based solution being a tool for you at this point. Send your comments publicly or privately - let me know if you want to remain anonymous, otherwise I'll give you free advertising by using your name or company name in my talk if I use your story. PS: Of course, the talk/slides will be available with Creative Commons Attribution-Noncommercial-Share Alike 3.0 United States License, and I expect that I'll probably use all or parts of this talk a few times this year, given the focus on the economy. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories needed
On Wed, 18 Feb 2009 13:37:57 John Todd wrote: I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on Sunday in Los Angeles, and the topic of my talk is Open Source in an Economic Downturn. I've got lots of talking points for this talk, but it would be interesting to hear some short anecdotes about how you in the Asterisk community are thriving, or at least surviving, by virtue of the benefits of Open Source. I find that real-world examples are worth more than all of the bullet points in the world, and timely stories from the community would be more interesting than hearing me prattle on. What economic downturn? I'm sick and tired of hearing this mantra. Michael ___ -- 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
