Re: [asterisk-users] asterisk fails when DNS or internet fails
I have setup a reverse dns for my local subnet and it seems to have resolved the issue, i was able to make calls even when my asterisk box is not connected to the net. thanks for all your help! On Wed, Jun 1, 2011 at 5:57 AM, Hans Witvliet h...@a-domani.nl wrote: On Tue, 2011-05-31 at 10:29 -0400, Eric Wieling wrote: As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. I'm not sure if that's all is true. Sure, if you add a line in /etc/hosts, that works for most applications, as not all commands follow /etc/resolv.conf i just tried, adding a line to /etc/hosts. ping hostname works, but host hostname fails, just as host ip-address. So even when you only put ip-addresses (brrr) into your config files, the reversed-lookup will still spoil the party. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
may i know what domain is asterisk specifically looking for? coz i don't use domains on the ip phones, i configure them to register to the IP e.g. 10.10.10.1. forgot to mention i am using freepbx as a GUI, does freepbx tells asterisk to look for a specific domain? TIA. Regards, Ron On Tue, May 31, 2011 at 9:17 PM, Olle E Johansson o...@edvina.net wrote: 31 maj 2011 kl. 14.49 skrev Benny Amorsen: Jeff LaCoursiere j...@sunfone.com writes: Hasn't anyone managed to solve this with something better than a caching DNS server, which seems to only last a short while? What exactly is going on that is failing? If your recursive DNS server returns errors quickly rather than actually trying to look up the names, Asterisk works fine. It is not a particularly nice workaround, but it does work... As long as Asterisk does not actually NEED the DNS information, but that can be most worked around with static configuration of IP addresses in sip.conf. Longterm we should really integrate an Asynchronus DNS library, like C-Ares. I've been wanting to do that for years. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
thank you eric. i have setup a reverse dns for my internal IP, hopefully that works. thanks again! regards Ron On Tue, May 31, 2011 at 10:29 PM, Eric Wieling ewiel...@nyigc.com wrote: As far as I can tell it is trying to do a reverse lookup on the IPs configured on the system. With the internet down, does the command host 10.10.10.1 (or whatever IPs you have on the system) take a while to come back? Unless you can do a reverse lookup of all the IPs on the system don't expect Asterisk to be able to. If your /etc/hosts is set up correct, you should be able to look up any IP configured on any interface on the system without delay. I'm sure there are other places Asterisk tries to do DNS lookups, but the above info has solved this issue for me in the past. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nhadie ramos Sent: Tuesday, May 31, 2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk fails when DNS or internet fails may i know what domain is asterisk specifically looking for? coz i don't use domains on the ip phones, i configure them to register to the IP e.g. 10.10.10.1. forgot to mention i am using freepbx as a GUI, does freepbx tells asterisk to look for a specific domain? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
Thank you for the information. I will try to install a dns-cache. Regards, Ron On 5/30/11, Alex Balashov abalas...@evaristesys.com wrote: On 05/30/2011 02:44 AM, gincantalupo wrote: - do not use urls, only ip addresses in sip.conf or put your urls inside /etc/hosts (is what I do especially sip providers urls) Definitely don't put URLs in /etc/hosts. I assume you meant URIs, but either way, neither one belongs there. That file is for overriding what would otherwise be a remote DNS query for a singular host (for applications using the libc resolver), and should only contain hostnames, short hostnames and IP addresses. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
By the way, is this only an issue for asterisk 1.4? or is it the same with 1.6 and/or 1.8? TIA. Regards, Ron On Mon, May 30, 2011 at 2:50 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/30/2011 02:44 AM, gincantalupo wrote: - do not use urls, only ip addresses in sip.conf or put your urls inside /etc/hosts (is what I do especially sip providers urls) Definitely don't put URLs in /etc/hosts. I assume you meant URIs, but either way, neither one belongs there. That file is for overriding what would otherwise be a remote DNS query for a singular host (for applications using the libc resolver), and should only contain hostnames, short hostnames and IP addresses. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk fails when DNS or internet fails
Hi, Would just like to inquire why asterisk fails to send calls in / out when the DNS is failing or when the server with asterisk has no internet. Ip phones are connected via IP address and i am using an FXO card, so even if internet fails i should still be able to make calls thru the fxo. but that is not the case. i am using the latest asterisk 1.4 and dahdi. any help would truly be appreciated. thank you. Regards, Ron -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sometimes extensions can't be called
Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes extensions can't be called
Hi, i see my extensions are there: 118103/118103 210.212.213.214 D N 5060 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118103/118103 210.212.213.214 D N 5060 Unmonitored and i have this on both servers: 17 sip peers [Monitored: 0 online, 0 offline Unmonitored: 15 online, 2 offline] regards, nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conference bridge
Hi, How can i setup conference when i have 2 asterisk servers? my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just for redundancy (not really high availability). i have a web interface, wherein i can create extension, conference etc. adding extension is ok, even if ext1 is registered on Asterisk 1 and ext2 is registered on asterisk 2 they will still be able to call each other, but on the conference, e.g. when ext1 dials conference no. 1000 and ext 2 dials conf 1000 also, they will be connected to two different conference room. my meetme is also setup on realtime. how can i set it up in such a way ext on registered on different asterisk server can connect to the same conference room. Regrdas, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AS5400 E1 SS7
Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI. this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
Hi, I don't know what i'm doing wrong but i already reinstalled the system. still using ubuntu 64-bit. made sure i had the correct local date time. then did all this: ntpdate pool.ntp.org tzselect , i chose Asia/SIngapore /etc/timezone is Asia/Singapore i added TZ='Asia/Singapore'; export TZ to /etc/profile. date shows the correct date, i rebooted, date still shows the correct date. then installed zaptel, ./configure, make menuselect, make make install make config then libpri make make install then asterisk, ./configure, make menuselect, make, make install, make samples then asterisk-addons, ./configure make menuselect, make make install make samples then run asterisk, then connect via asterisk -r [Jun 13 00:25:58] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found [Jun 13 00:26:02] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found log shows June 13 00:25 my system date shows /home/ronald# date Fri Jun 13 15:33:03 SGT 2008 i installed everything as root via sudo su, how come i still dont get the correct time? really need help on this one. thank you regards nhadie --- On Fri, 6/13/08, Lee, John (Sydney) [EMAIL PROTECTED] wrote: From: Lee, John (Sydney) [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, June 13, 2008, 1:22 AM i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. You should actually be using Asia/Singapure rather than guess. i'm not using ntp, coz when i do i also don't get the correct time. That's because you have an incorrect timezone set. I am also using gotoiftime in my IVR but I don't have any problems. 1) Install the distro and specify the timezone 2) Set the correct time in linux 3) Install ntp 4) Sync the time by ntpdate ntp will always just sync using GMT time but the timezone specified in the distro will provide the time difference and daylight savings. That is it! Also, can someone clarify if Asterisk really uses a different time than the system time? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
Hi Sir, what i did is reinstall (again) but this time using debian 32-bit. and now i get the time correctly. so i'm not sure if it's a prob with ubuntu or asterisk, or asterisk on ubuntu, or asterisk on ubuntu 64-bit. coz i dont know how to figure those out. but anyway debian+asterisk works fine. thanks to all your reply! regards ron --- On Fri, 6/13/08, Tilghman Lesher [EMAIL PROTECTED] wrote: From: Tilghman Lesher [EMAIL PROTECTED] Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, June 13, 2008, 12:31 PM On Friday 13 June 2008 02:35:09 Nhadie Ramos wrote: Hi, I don't know what i'm doing wrong but i already reinstalled the system. still using ubuntu 64-bit. made sure i had the correct local date time. then did all this: ntpdate pool.ntp.org tzselect , i chose Asia/SIngapore /etc/timezone is Asia/Singapore i added TZ='Asia/Singapore'; export TZ to /etc/profile. date shows the correct date, i rebooted, date still shows the correct date. then installed zaptel, ./configure, make menuselect, make make install make config then libpri make make install then asterisk, ./configure, make menuselect, make, make install, make samples then asterisk-addons, ./configure make menuselect, make make install make samples then run asterisk, then connect via asterisk -r [Jun 13 00:25:58] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found [Jun 13 00:26:02] NOTICE[5159]: chan_sip.c:15075 handle_request_register: Registration from '11002 sip:[EMAIL PROTECTED]' failed for '202..156.117.155' - No matching peer found log shows June 13 00:25 my system date shows /home/ronald# date Fri Jun 13 15:33:03 SGT 2008 i installed everything as root via sudo su, how come i still dont get the correct time? The only thing I can think of is that your zoneinfo files are not in the right place. Does the file /usr/share/zoneinfo/Asia/Singapore exist? Also, does a symlink exist from that file to /etc/localtime? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
Hi Sir, I tried restarting asterisk, but still it has the wrong time. I tried restarting the system, then start asterisk it still uses the wrong time. I also tried recompiling asterisk, checked i have the correct time on the system,nbsp; then restart the system then start asterisk but still i get the wrong time. My system time (currently) Thu Jun 12 15:12:11 GST 2008 on asterisk i use EPOCH to look at the time, nbsp; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) i would really appreciate any help. TIA ron --- On Thu, 6/12/08, Tilghman Lesher lt;[EMAIL PROTECTED]gt; wrote: From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 1:42 AM On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; between the asterisk time and the system time. could it be because i gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; of timezone on the system? How can I fix this prob? Yes, that's probably the reason. The system timezone is cached once at startup, for performance reasons. The only way to get it to pick up the new timezone is a restart. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 lt;[EMAIL PROTECTED]gt; wrote: From: mkn0014 lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: gt; Hi Sir, gt; gt; I tried restarting asterisk, but still it has the wrong time. gt; gt; I tried restarting the system, then start asterisk it still uses the gt; wrong time. gt; gt; I also tried recompiling asterisk, checked i have the correct time on gt; the system, then restart the system then start asterisk but still i gt; get the wrong time. gt; gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008 gt; gt; on asterisk i use EPOCH to look at the time, gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) gt; gt; i would really appreciate any help. TIA gt; gt; ron gt; gt; --- On *Thu, 6/12/08, Tilghman Lesher gt; /lt;[EMAIL PROTECTED]gt;/* wrote: gt; gt; From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; gt; Subject: Re: [asterisk-users] time on asterisk gt; To: Asterisk Users Mailing List - Non-Commercial Discussion gt; lt;asterisk-users@lists.digium.comgt; gt; Date: Thursday, June 12, 2008, 1:42 AM gt; gt; On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; gt; between the asterisk time and the system time. could it be because i gt; gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; gt; of timezone on the system? How can I fix this prob? gt; gt; Yes, that's probably the reason. The system timezone is cached once at gt; startup, for performance reasons. The only way to get it to pick up the new gt; timezone is a restart. gt; gt; -- gt; Tilghman gt; gt; Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] time on asterisk
hi sir, i forgot to mention it was originally at Asia/Singapore, when i noticed that asterisk has a wrong time, that's why i tried GMT instead. regards, Ron --- On Thu, 6/12/08, Stelios Koroneos lt;[EMAIL PROTECTED]gt; wrote: From: Stelios Koroneos lt;[EMAIL PROTECTED]gt; Subject: RE: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 9:33 AM GMT timezone does not have daylight savings, so probably this is why you have the wrong time Select a timezone for a city and usually the correct daylight parameters are used nbsp; Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com nbsp; From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Thursday, June 12, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] time on asterisk hi mats, i'm using 64-bit Ubuntu Server Edition 8.04 I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but if i use GMT+8 the system does not give the correct time. i'm not using ntp, coz when i do i also don't get the correct time. i'm not sure how i can fix this, is this an ubuntu issue? regards, ron --- On Thu, 6/12/08, mkn0014 lt;[EMAIL PROTECTED]gt; wrote: From: mkn0014 lt;[EMAIL PROTECTED]gt; Subject: Re: [asterisk-users] time on asterisk To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion lt;asterisk-users@lists.digium.comgt; Date: Thursday, June 12, 2008, 8:20 AM Nhadie Ramos wrote: gt; Hi Sir, gt; gt; I tried restarting asterisk, but still it has the wrong time. gt; gt; I tried restarting the system, then start asterisk it still uses the gt; wrong time. gt; gt; I also tried recompiling asterisk, checked i have the correct time on gt; the system, then restart the system then start asterisk but still i gt; get the wrong time. gt; gt; My system time (currently) Thu Jun 12 15:12:11 GST 2008 gt; gt; on asterisk i use EPOCH to look at the time, gt; NoOp(SIP/105101-00857e60, DATE: 20080612-081147) gt; gt; i would really appreciate any help. TIA gt; gt; ron gt; gt; --- On *Thu, 6/12/08, Tilghman Lesher gt; /lt;[EMAIL PROTECTED]gt;/* wrote: gt; gt; From: Tilghman Lesher lt;[EMAIL PROTECTED]gt; gt; Subject: Re: [asterisk-users] time on asterisk gt; To: Asterisk Users Mailing List - Non-Commercial Discussion gt; lt;asterisk-users@lists.digium.comgt; gt; Date: Thursday, June 12, 2008, 1:42 AM gt; gt; On Wednesday 11 June 2008 17:52:15 Nhadie Ramos wrote: gt; gt; I'm using gotoiftime on asterisk, but it seemsamp;nbsp; there is a difference gt; gt; between the asterisk time and the system time. could it be because i gt; gt; adjusted the system timezone on my linux? do asterisk not detect the change gt; gt; of timezone on the system? How can I fix this prob? gt; gt; Yes, that's probably the reason. The system timezone is cached once at gt; startup, for performance reasons. The only way to get it to pick up the new gt; timezone is a restart. gt; gt; -- gt; Tilghman gt; gt; Ron, What OS/Distro are you using ? What timezone are you using ? Do you use NTP for syncing time/date? /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] time on asterisk
Hi, I'm using gotoiftime on asterisk, but it seemsnbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie ___ -- Bandwidth and Colocation Provided 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] Follow-up Question Was: Question on DeadAGI
Hi,
i noticed a alot of mistake on what i did.
i have this macro
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,SetMusicOnHold(${ARG3})
exten =gt; s,Set(TIMEOUT(absolute)=${ARG4})
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n,Hangup()
exten =gt; h,1,ResetCDR(w)
exten =gt; h,n,NoCDR()
exten =gt; h,n,DEADAGI(get-total.php)
[outbound-trunk-100]
exten =gt; _00.,1,AGI(call-compute.php)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 1]?80)
exten =gt; _00.,n,Hangup
exten =gt; _00.,80,Macro(dialout-trunk|${EXTEN}|intl-trunk|moh-100|${OUTTIME})
exten =gt; _00.,n,Hangup
I tried calling my mobile , call-compute.php was executed,i'm able to see
details i need for start accounting.
When i answer my phone and hangup, get-total is executed also.
My prob is ifnbsp; i cancel my the call on my mobile, ip phone keeps on
dialing it. How can i detect that the other end canceled the call?
Another is if i dial any number, even invalid ones, my script get-total.php
still thinks it is an answered call, so it still does deducting on the balance.
will really appreciate any help.nbsp; TIA.
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: asterisk-users@lists.digium.com
Date: Saturday, June 7, 2008, 10:52 PM
Thanks to all the help. I think i have it now. I reset the CDR on the hangup
channel.
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n,Hangup()
exten =gt; h,1,ResetCDR(w)
exten =gt; h,n,NoCDR()
exten =gt; h,n,DEADAGI(get-total.php)
AGI Rx lt;lt; EXEC Noop ROWCOUNT=1
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(ROWCOUNT=1)
AGI Tx gt;gt; 200 result=0
AGI Rx lt;lt; EXEC Noop BILLSEC=21
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(BILLSEC=21)
now i can see my billsec. thanks again for all the help.
regards,
nhadie
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 10:39 PM
Hi Sir,
I tried it this way, and now i can see my DEADGI being called next prob is
onnbsp; that script i query the cdr table with the uniqueid. tried counting
the row result first , and result was 0.
how can i make sure that it was already at the CDR table before i call my agi?
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the
row.
but when i query manully on the mysql console, i can see the cll was logged.
Thank You
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n.ResetCDR()
exten =gt; s,n,Hangup
exten =gt; h,1,DEADAGI(get-total.php)
---
On Sat, 6/7/08, Lenz
lt;[EMAIL PROTECTED]gt; wrote:
From: Lenz lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt;
gt; but after the call i would like to get the usage by calling
gt; get-usage.php so
i
can update users balance, but looking at the debug it
gt; seems the AGI was not called. is there som
gt;
gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
1]?70)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
2]?80)
gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt; exten =amp;gt; _00.,n,Hangup
gt; exten =amp;gt; _00.,n,DEADAGI(get-usage.php)
gt; exten =amp;gt; _00.,80,Busy
gt; exten =amp;gt; _00.,n,Hangup
gt;
gt;
gt; Regards,
gt; Nhadie
gt;
gt;
gt;
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided 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
Re: [asterisk-users] Question on DeadAGI
Hi sir,
i'm sorry but how can i use it on a hangup channel?
gt;gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt;gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} = 1]?70)
gt;gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} = 2]?80)
gt;gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt;gt; exten =amp;gt; _00.,n,Hangup
gt;gt; exten =amp;gt; h,n,DEADAGI(get-usage.php)nbsp;nbsp;
lt;---should i do it like this?
gt;gt; exten =amp;gt; _00.,80,Busy
gt;gt; exten =amp;gt; _00.,n,Hangup
thank you
--- On Sat, 6/7/08, Lenz lt;[EMAIL PROTECTED]gt; wrote:
From: Lenz lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt;
gt; but after the call i would like to get the usage by calling
gt; get-usage.php so i can update users balance, but looking at the debug it
gt; seems the AGI was not called. is there som
gt;
gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
1]?70)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
2]?80)
gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt; exten =amp;gt; _00.,n,Hangup
gt; exten =amp;gt; _00.,n,DEADAGI(get-usage.php)
gt; exten =amp;gt; _00.,80,Busy
gt; exten =amp;gt; _00.,n,Hangup
gt;
gt;
gt; Regards,
gt; Nhadie
gt;
gt;
gt;
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided 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 on DeadAGI
Hi Sir,
I tried it this way, and now i can see my DEADGI being called next prob is
onnbsp; that script i query the cdr table with the uniqueid. tried counting
the row result first , and result was 0.
how can i make sure that it was already at the CDR table before i call my agi?
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the
row.
but when i query manully on the mysql console, i can see the cll was logged.
Thank You
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n.ResetCDR()
exten =gt; s,n,Hangup
exten =gt; h,1,DEADAGI(get-total.php)
--- On Sat, 6/7/08, Lenz lt;[EMAIL PROTECTED]gt; wrote:
From: Lenz lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt;
gt; but after the call i would like to get the usage by calling
gt; get-usage.php so i can update users balance, but looking at the debug it
gt; seems the AGI was not called. is there som
gt;
gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
1]?70)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
2]?80)
gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt; exten =amp;gt; _00.,n,Hangup
gt; exten =amp;gt; _00.,n,DEADAGI(get-usage.php)
gt; exten =amp;gt; _00.,80,Busy
gt; exten =amp;gt; _00.,n,Hangup
gt;
gt;
gt; Regards,
gt; Nhadie
gt;
gt;
gt;
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided 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 on DeadAGI
Thanks to all the help. I think i have it now. I reset the CDR on the hangup
channel.
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n,Hangup()
exten =gt; h,1,ResetCDR(w)
exten =gt; h,n,NoCDR()
exten =gt; h,n,DEADAGI(get-total.php)
AGI Rx lt;lt; EXEC Noop ROWCOUNT=1
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(ROWCOUNT=1)
AGI Tx gt;gt; 200 result=0
AGI Rx lt;lt; EXEC Noop BILLSEC=21
nbsp;nbsp;nbsp; -- AGI Script Executing Application: (Noop) Options:
(BILLSEC=21)
now i can see my billsec. thanks again for all the help.
regards,
nhadie
--- On Sat, 6/7/08, Nhadie Ramos lt;[EMAIL PROTECTED]gt; wrote:
From: Nhadie Ramos lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 10:39 PM
Hi Sir,
I tried it this way, and now i can see my DEADGI being called next prob is
onnbsp; that script i query the cdr table with the uniqueid. tried counting
the row result first , and result was 0.
how can i make sure that it was already at the CDR table before i call my agi?
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the
row.
but when i query manully on the mysql console, i can see the cll was logged.
Thank You
[macro-dialout-trunk]
exten =gt; s,1,Wait(1)
exten =gt; s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten =gt; s,n.ResetCDR()
exten =gt; s,n,Hangup
exten =gt; h,1,DEADAGI(get-total.php)
--- On Sat, 6/7/08, Lenz
lt;[EMAIL PROTECTED]gt; wrote:
From: Lenz lt;[EMAIL PROTECTED]gt;
Subject: Re: [asterisk-users] Question on DeadAGI
To: Asterisk Users Mailing List - Non-Commercial Discussion
lt;asterisk-users@lists.digium.comgt;
Date: Saturday, June 7, 2008, 12:50 PM
You should use it on the hang-up extension and only after the channel is
technically dead.
It works fine for that.
l.
On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos lt;[EMAIL PROTECTED]gt;
wrote:
gt; Hi,
gt;
gt; How can i get the deadAGI to work at this scenario
gt;
gt; Basically when someonc calls international,amp;nbsp; i will get the
gt; remaining balance using AGI get-available.php.
gt;
gt; but after the call i would like to get the usage by calling
gt; get-usage.php so i
can update users balance, but looking at the debug it
gt; seems the AGI was not called. is there som
gt;
gt; exten =amp;gt; _00.,1,AGI(get-available.php)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
1]?70)
gt; exten =amp;gt; _00.,n,GotoIf($[${CALLSTATUS} =
2]?80)
gt; exten =amp;gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
gt; exten =amp;gt; _00.,n,Hangup
gt; exten =amp;gt; _00.,n,DEADAGI(get-usage.php)
gt; exten =amp;gt; _00.,80,Busy
gt; exten =amp;gt; _00.,n,Hangup
gt;
gt;
gt; Regards,
gt; Nhadie
gt;
gt;
gt;
--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided 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 on DeadAGI
Hi,
How can i get the deadAGI to work at this scenario
Basically when someonc calls international,nbsp; i will get the remaining
balance using AGI get-available.php.
but after the call i would like to get the usage by calling get-usage.php so i
can update users balance, but looking at the debug it seems the AGI was not
called. is there som
exten =gt; _00.,1,AGI(get-available.php)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 1]?70)
exten =gt; _00.,n,GotoIf($[${CALLSTATUS} = 2]?80)
exten =gt; _00.,70,Dial(SIP/[EMAIL PROTECTED])
exten =gt; _00.,n,Hangup
exten =gt; _00.,n,DEADAGI(get-usage.php)
exten =gt; _00.,80,Busy
exten =gt; _00.,n,Hangup
Regards,
Nhadie
___
-- Bandwidth and Colocation Provided 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] No Audio on Meetme
Hi All, What could be the cause why there is no audio coming form the participants. ztdummy is loaded, ZTDUMMY/1 (source: HRtimer) 1. I can hear Please enter your PIN, User blah blah has enttered...etc etc But when the particpants talk, we hear nothing. What are the possible mistakes i did on these? TIA Nhadie ___ -- Bandwidth and Colocation Provided 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] ser to load balance asterisk
Hi,
i will try to setup 3 * box, 1 ser.
if none, let's say i have 4 extensions 101, 102,103 and 104, 101 registered on
* 1, 102 on * 2, 103 on * 3 and 104 on * 1 also.
i will define this dial plan:
[dial-extension]
exten = _1XX,1,Dial(SIP/${EXTEN}) - look it up on the local first
exten = _1XX,1,Dial(SIP/[EMAIL PROTECTED]) - if not on local, check in ser
when ser receives a request, it will ask the other 2 * servers where the
request did not come from, e.g if call originated from ask * 1 first if not
then ask * 2. Do you think there will be issues?
i also plan on installing heartbeat so users will only register on a single
host., or can i simply use DNS SRV? is my setup possible? TIA
regards,
nhadie
-
Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___
-- Bandwidth and Colocation Provided 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] meetme hungs up
Hi all, got some issues with meetme, i created meetme 8000, i have users 200 and 201 dial to 8000, meetme works fine. but i need an outside user to join, so that user will dial-in via the sip trunk. sip trunk has did e.g. , outside user dials-in to that DID, asterisk rcvs it forward it to 8000. meetme prompt will play asking for pin etc etc. once connected to the meetme, after 1-2 secs the outside user will be disconnected from the conference. but local extension 200 and 201 is still ok. during disconnect i saw this on the logs, Hungup 'Zap/pseudo-1147226235' how can i fix these prob? TIA. regards, nhadie - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided 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] prepaid on the trunks
if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use the trunk as a unique identifier for each customer not their local extension. e.g. customer A will have trunk 34587612 and will have extension 101 and 102, customer B will have a different trunk 87659043 but will also have the extension 101 and 102. i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? thank you. regards, nhadie - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided 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] prepaid on the trunks
Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Regards, Nhadie On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote: i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? TBH, I don't really understand your description, but I will say that I implemented astcc a week or two ago and it works for what I need. Cheers, b. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided 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] prepaid on the trunks
hi sir, yes that would be it, but instead of having a prepaid provider, i will setup my own as5300 and asterisk will talk to that. is that possible in astcc, astbill or a2billing? regards, nhadie Am I correct in thinking that one application of this would be monitoring what you have left for funds with a prepaid vendor? Darren Wiebe [EMAIL PROTECTED] Brian J. Murrell wrote: On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Wouldn't you just equate a Calling Card (that's the unit that has an account balance and charges against it) with a trunk instead of a user or extension? You can call the astcc agi script with any value you want for a Calling Card identifier. b. Nhadie Ramos [EMAIL PROTECTED] wrote: Hi, sorry to confused you with my question. the normal prepaid application like astcc, if i'm not mistaken, monitors the amount left on the user (which i usually refer as extension), what i want to do is monitor prepaid on the trunk (or the SIP channel use to call outbound to pstn). Is that possible? Regards, Nhadie On Tue, 2008-04-22 at 22:59 -0700, Nhadie Ramos wrote: i want to create a billing system to monitor only the trunks and also to load amounts on those trunks. is this possible? will i be able to use app_prepaid for this? TBH, I don't really understand your description, but I will say that I implemented astcc a week or two ago and it works for what I need. Cheers, b. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided 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] prepaid on the trunks
thank you sir, i will try to check on that. i haven't really tried astcc yet so
i really dont understand how it works right now.
also, do you have any reference on using app_prepaid? can't find some sample
config, i would like to see how i can use that. do you think app_prepaid is
suited for the scenario i have?
thank you
regards
nhadie
Brian J. Murrell [EMAIL PROTECTED] wrote: On Wed, 2008-04-23 at 15:41
-0600, Darren Wiebe wrote:
Ok, I'm not aware of this feature in astcc
Keep in mind that astcc is simply a tool that keeps a database of
minutes used for some entity (typically a calling card) and calculates
those minutes used against a pre-charged amount. The number of the
entity can be passed to astcc (i.e. so that it does not need to prompt
the user for it) in such a way:
exten = _1NXXNXX,n,DeadAGI(astcc.agi,${cardnum},${EXTEN})
So binding a trunk to a cardnum (i.e. a given pre-charged account)
should be easy enough to do.
b.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-
Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___
-- Bandwidth and Colocation Provided 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] meetme with time condition
Hi All, How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) regards, nhadie ___ -- Bandwidth and Colocation Provided 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] meetme with time condition
Hi All, How can i enable time condition on meetme? below i would like to deny callers if the time is not yet the scheduled time of the conference, but it seems like its still goes to 600,2, hope anyone can help. [meet-me-test] exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3) exten = 600,2,Playback(vm-goodbye) exten = 600,3,Hangup exten = 600,4,MeetMe(600||600600) regards, nhadie - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- Bandwidth and Colocation Provided 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] meetme with time condition
Hi Steve,
Thanks for the reply. I made it look like this:
[meet-me-test]
exten = 600,1,GotoIfTime(11:00-12:00|*|20|Apr?meetnow)
exten = 600,n,Answer
exten = 600,n,Festival('Conference is not yet active, please dial in on the
assigned time');
exten = 600,n,Hangup
exten = 600,n(meetnow),MeetMe(600||600600)
exten = 600,n,Hangup
and it's kind of working now, except for the Festival I added, i dont hear any
audio but it's executing it
Executing [EMAIL PROTECTED]:3] Festival(SIP/1100-b691a738, Conference is
not yet active, please dial in on the assigned time) in new stack.
also, how can i limit the time of the conference? on this scenario how would it
get cut on 12:00? is it also possible to extend the time limit even if the
conference is already ongoing? thanks alot.
regards,
nhadie
=
1) I read this as If the time is between 10 and 11 in the morning on the
19th of April, hangup. Else, play goodbye and hangup. Neither case will
get you to the conference.
2) Is it still the 19'th in your part of the world?
3) Try replacing each element with * to identify the source of your
problem.
And a few suggestions:
1) Since the target of your gotoiftime is in the same context and the same
extension, you only need to specify the priority.
2) You can use the n priority.
Both will make your code more robust, maintainable, and reusable.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
Nhadie Ramos [EMAIL PROTECTED] wrote: Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
exten = 600,2,Playback(vm-goodbye)
exten = 600,3,Hangup
exten = 600,4,MeetMe(600||600600)
regards,
nhadie
-
Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it
now.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-
Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___
-- Bandwidth and Colocation Provided 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] voicemail odbc storage
Hi, I was able to store voicemail following the tutorial http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage i would just like to inquire how can i create a web interface (will use php) to play the voicemail stored in the database. the field in the database is recording longblob anyone able to retrieve that file and play on an embedded player on a webpage? Thank You Regards Nhadie ___ -- Bandwidth and Colocation Provided 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] features on dial pad
Hi All, If i were to develop a softphone, how can i add call transfer, call on hold and 3-way conference on it? linksys Ip phone has those built-in button to xfer, conf, on hold. and x-lite also has those, how can i have those if i develop my own? Thank You Regards, Nhadie ___ -- Bandwidth and Colocation Provided 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] Outbund Route via Extension
Hi All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source. i'm using freepbx for this. thank you Nhadie ___ --Bandwidth and Colocation Provided 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] :THIS IS A SPAM: Re: Sangoma on Fedora 7 x86_64
Hi john, Thank you for your reply, i finally stumbled on google what the problem is. The driver does not compile on kernel newer than 2.6.19. Regards, Nhadie John Novack wrote: Sangoma gives EXCELLENT technical support. I would suggest you try there first. The few problems I have had with installation were addressed promptly and when driver fixes proved necessary, corrected in short order. Also the cards have a 5 year warranty! John Novack ___ --Bandwidth and Colocation Provided 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] Sangoma on Fedora 7 x86_64
Hi, I'm trying to install asterisk(v1.2.22) with FreepBX(v2.2.3) with a 4-Port FXO Sangoma card A200. I'm using Fedora 7 (x86_64) kernel version 2.6.22.1-27.fc7, but i'm having these errors: $ ztcfg - Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) $ lspci -v 02:01.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: NEC Corporation Unknown device 1000 Flags: bus master, medium devsel, latency 64, IRQ 10 Memory at fdde (32-bit, non-prefetchable) [size=64K] Nothing else uses IRQ 10. An error when i installed wanpipe stable version 2.3.4-12, i also get the same error when i used wanpipe 3.1.2 WANPIPE DRIVER COMPILE LOG Thu Jul 26 21:26:33 PHT 2007 --- make -C /lib/modules/2.6.22.1-27.fc7/build SUBDIRS=/usr/local/src/wanpipe-2.3.4-12/kdrvtmp CC=gcc KBUILD_VERBOSE=0 modules make[1]: Entering directory `/usr/src/kernels/2.6.22.1-27.fc7-x86_64' CC [M] /usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.o In file included from /usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.c:135: include/linux/wanpipe_common.h: In function ‘wan_skb_tail’: include/linux/wanpipe_common.h:1017: warning: return makes pointer from integer without a cast include/linux/wanpipe_common.h: In function ‘wan_skb_set_raw’: include/linux/wanpipe_common.h:1281: error: ‘struct sk_buff’ has no member named ‘mac’ include/linux/wanpipe_common.h:1282: error: ‘struct sk_buff’ has no member named ‘nh’ include/linux/wanpipe_common.h: In function ‘wan_skb_init’: include/linux/wanpipe_common.h:1735: warning: assignment makes integer from pointer without a cast make[2]: *** [/usr/local/src/wanpipe-2.3.4-12/kdrvtmp/sdladrv_src.o] Error 1 make[1]: *** [_module_/usr/local/src/wanpipe-2.3.4-12/kdrvtmp] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.22.1-27.fc7-x86_64' make: *** [all] Error 2 That error i really don't understand. Has anybody tried to install Sangoma on Fedora 7? TIA Ronald ___ --Bandwidth and Colocation Provided 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] SER local as an Asterisk Trunk
Hi, Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk, i used an SER local as a trunk for the Asterisk. When the Asterisk box register to SER it will have this URI sip:[EMAIL PROTECTED], instead of sip:[EMAIL PROTECTED] Anyone has encountered this problem? Because I'm checking the From part, and s is not a valid extension number so it will deny it calling to the gateway. TIA Regards Nhadie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
