Re: [asterisk-users] This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
Zeeshan Zakaria ha scritto: I have two test asterisk boxes, both version 1.4.26, on which I do Answer() followed by MusicOnHold() and it works just fine. I do this all the time as this is my standard way of testing new contexts. Yesterday i tested another installation and i found the same issue. Maybe the problem is SIP related or console channel related. I explain (if someone can do a test i am happy). Go to the asterisk console, place a dial command calling thru the SIP trunk, then place a transfer to the extension MusicOnHold after the Answer... (this is the sequence) dial 0num...@from-sip (the from-sip is the context where a sip phone can dial to the trunk) pick up the phone called transfer *...@from-sip (the *199 extension is Answer - MusicOnHold) you must hear the music on the phone called (or not) So this may be a console channel problem... Yesterday i try to use the outgoing spool (place a file on /var/spool/asterisk/outgoing making a call to the phone and directly go to the *199 extension, the same thing i do on console automated with no console channel), audio ok. So i am going to open a bug... :-) Thnks. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-07-07 4:16 AM, Massimo Nuvoli mass...@archivio.it mailto:mass...@archivio.it wrote: I found a strange thing on Asterisk 1.4.32, the same defect on 1.4.26.? I spend 4 hours to try to solve... but found only a workaround. As is easy to reproduce the problem i need to know if this is a bug or if there is some idiot configuration that i miss. Maybe also the bug is know... Scenario: Asterisk installation on ubuntu 9.04 64 bit. Trunk SIP (two different providers) On the Asterisk server there are a number of SIP clients. If i use the sip client all things ok, i made a call and everything ok. If i place the call from the server (or if i call trhu the SIP trunk the asterisk system) everytime the Answer() application seeems to NOT work. The only way to make it work is to use some other function that do the Answer in place. (call coming from the SIP trunk) If i use Answer() MusicOnHold() I hear nothing. If i use Answer() PlayBack(silence/1) MusicOnHold() or Answer() VoiceMail(1...@default) i can hear all ok (it seems that the PlayBack and the VoiceMail apps are able to Answer really...) I checked the SIP debug trace, it seems no problem on the SIP side. Thnks guys. -- _ -- 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 attachment: massimo.vcf signature.asc Description: OpenPGP digital signature -- _ -- 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] Not detecting hangup
That looks like the option that will help a lot. Thanks. On 8 July 2010 23:21, Steve Edwards asterisk@sedwards.com wrote: [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? On Thu, 8 Jul 2010, Danny Nicholas wrote: First thought is that you can put a timeout on your calls, but that is just a band-aid. Also not fixing the source of the problem, but rtpholdtimeout and rtptimeout may help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. I appreciate your advice. Thanks, Bruce -- _ -- 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] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
bruce bruce ha scritto: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. Hu interesting... i never checked this kind of product. CyberData has calculator only for the 8W model, but... every speaker they sell is 8w and the calculator say 69 speakers. You can attach 2 speakers to one amplifier in parallel (they say this also), this is the maximum as the amplifier cannot reach 32W (4 speakers), but you can try to use 4 speakers (2 in parallel + 2 in parallel) with a little less than maximum 8w on each speaker. I think a reasonable number of speakers may be less than half, but you must check wath is in the area, also remember if this is a warehouse to place the speakers where a person can be, not goods. :-) For a so big installation think to use a voip interface and professional product with low voltage line speakers, i think is less expensive. Bye. attachment: massimo.vcf signature.asc Description: OpenPGP digital signature -- _ -- 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] [NAT] * + private IP + locked-down firewalls?
On Mon, 05 Jul 2010 12:45:34 +0200, Gilles codecompl...@free.fr wrote: Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP... For those interested, I was tipped through private e-mail about using OpenVPN to open a steady tunnel between the client and Asterisk, and have the SIP client send packets through that tunnel instead of trying to connect directly. -- _ -- 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] Click2call from an OpenOffice document
Hi, What would you suggest to get click2call from an OpenOffice document ? For instance, in OOo Writer, there is a block : M. John Doe Tel: +1 234 567 890 email: j...@example.com Looking at this block, the line +1 234 567 890 is underlined. When clicking on this, a contextual menu pops up allowing you to make a call. Regards -- _ -- 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 Crashes - Segmentation Fault
Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x01027d9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh. On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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 and cisco 2800
Hi Peder, it seems to work, thank you! Now I've got a problem with the cisco 2800 which is resetting every 5 minutes but I do not think it is related to the cable, maybe something about the clock but except for a wiki page (http://www.voip-info.org/wiki/view/Asterisk+legacy+integration) there is nothing on internet about connecting asterisk and cisco... :( Giorgio Incantalupo Peder wrote: That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Tuesday, July 06, 2010 2:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 2800 Hi Neeraj, my problem is not terminating but making the Cisco accept the calls coming from my Asterisk. The bad news is I cannot have access to the Cisco sw, it is like a black box for me. The only thing I can have access to is the T1/E1 port on the back of the Cisco 2800. I made a custom cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c2e19c7.5090...@fgasoftware.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2 15:20:36] VERBOSE[15004] logger.c: -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new stack [Jul 2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Spawn extension (inbound, , 2) exited non-zero on 'IAX2/1-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- 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 Crashes - Segmentation Fault
Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x01027d9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh. On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.comwrote: Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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 Crashes - Segmentation Fault
It still crashes and in gdb trace following is what its showing: --More-- warning: .dynamic section for /usr/lib/mysql/libmysqlclient.so.15 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 13310] LAST FEW LINES IN GDB TRACE: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x003acd9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh On Fri, Jul 9, 2010 at 2:36 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi, Following is what i did. [r...@linuxtest ~]# yum install mysql* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centos.skknet.net * updates: centos.skknet.net Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and latest version Resolving Dependencies -- Running transaction check --- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated --- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated -- Finished Dependency Resolution Dependencies Resolved PackageArchVersionRepository Size Installing: mysql-benchi3865.0.77-4.el5_5.3 updates 507 k mysql-test i3865.0.77-4.el5_5.3 updates 3.7 M Transaction Summary Install 2 Package(s) Upgrade 0 Package(s) Total download size: 4.2 M Is this ok [y/N]: y Downloading Packages: (1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02 (2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm | 3.7 MB 00:11 Total 295 kB/s | 4.2 MB 00:14 Running rpm_check_debug Running Transaction Test Finished Transaction Test Transaction Test Succeeded Running Transaction Installing : mysql-bench 1/2 Installing : mysql-test 2/2 Installed: mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386 0:5.0.77-4.el5_5.3 Complete! [r...@linuxtest ~]# yum install mysql-devel* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centosr4.centos.org * updates: centosg4.centos.org Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Nothing to do [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf [r...@linuxtest ~]# vi /etc/ld.so.conf [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf /usr/lib/mysql [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# Thanks Regards Manmohan Singh On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Hi, Following is what i did. [r...@linuxtest ~]# yum install mysql* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centos.skknet.net * updates: centos.skknet.net Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-server-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-5.0.77-4.el5_5.3.i386 already installed and latest version Package mysql-connector-odbc-3.51.26r1127-1.el5.i386 already installed and latest version Resolving Dependencies -- Running transaction check --- Package mysql-bench.i386 0:5.0.77-4.el5_5.3 set to be updated --- Package mysql-test.i386 0:5.0.77-4.el5_5.3 set to be updated -- Finished Dependency Resolution Dependencies Resolved PackageArchVersionRepository Size Installing: mysql-benchi3865.0.77-4.el5_5.3 updates 507 k mysql-test i3865.0.77-4.el5_5.3 updates 3.7 M Transaction Summary Install 2 Package(s) Upgrade 0 Package(s) Total download size: 4.2 M Is this ok [y/N]: y Downloading Packages: (1/2): mysql-bench-5.0.77-4.el5_5.3.i386.rpm | 507 kB 00:02 (2/2): mysql-test-5.0.77-4.el5_5.3.i386.rpm | 3.7 MB 00:11 Total 295 kB/s | 4.2 MB 00:14 Running rpm_check_debug Running Transaction Test Finished Transaction Test Transaction Test Succeeded Running Transaction Installing : mysql-bench 1/2 Installing : mysql-test 2/2 Installed: mysql-bench.i386 0:5.0.77-4.el5_5.3 mysql-test.i386 0:5.0.77-4.el5_5.3 Complete! [r...@linuxtest ~]# yum install mysql-devel* Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: centos.skknet.net * base: centos.skknet.net * extras: centosr4.centos.org * updates: centosg4.centos.org Setting up Install Process Package mysql-devel-5.0.77-4.el5_5.3.i386 already installed and latest version Nothing to do [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf [r...@linuxtest ~]# vi /etc/ld.so.conf [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# cat /etc/ld.so.conf include ld.so.conf.d/*.conf /usr/lib/mysql [r...@linuxtest ~]# ldconfig [r...@linuxtest ~]# Thanks Regards Manmohan Singh On Fri, Jul 9, 2010 at 2:19 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg
Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?
Hi! Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP... For those interested, I was tipped through private e-mail about using OpenVPN to open a steady tunnel between the client and Asterisk, and have the SIP client send packets through that tunnel instead of trying to connect directly. Indeed - and both Snom and Yealink offer phones with OpenVPN support built-in. That way you do not need to worry about the router/firewall. BTW: I see that the Yealink firmware now also comes with 802.1x support. Am I correct that no other popular SIP phone brand has that? Philipp -- _ -- 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] AGI get full variable
Dear All, Please anyone help me to solve the following problem. Thanks, Velusamy On Thu, Jul 8, 2010 at 4:19 PM, velusamy Krishnan velu.techni...@gmail.comwrote: Dear All, I have get full variable AGI call to get the ANSWEREDTIME channel variable. I have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI GET FULL VARIABLE returns -1 for ANSWEREDTIME channel variable. What is the problem? Where I made wrong. Please suggest me.. Regards, Velusamy. -- _ -- 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] Re : Re : Re : Communication IAX2 SIPIAX2
ok it works i had a problem with a syntax:
i had to wrire:
exten =_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20))
thanks
De : Adil Zaaraoui adilzeaara...@yahoo.fr
À : Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Envoyé le : Jeu 8 juillet 2010, 19h 41min 15s
Objet : Re : [asterisk-users] Re : Communication IAX2 SIPIAX2
Yes i agree; ok here the output of verbosity at level 3:
-- Executing [00212664800...@pstn2:1] GotoIf(SIP/100-081e3648,
0?internal:external) in new stack
-- Goto (pstn2,00212664800450,2)
-- Executing [00212664800...@pstn2:2] Dial(SIP/100-081e3648,
SIP/lo...@pstn2/011212664800450||S(20)) in new stack
-- Setting call duration limit to 20 seconds.
[Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host:
pstn2/011212664800450
[Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/100-081e3648' status is 'CHANUNAVAIL'
-- Executing [...@pstn2:1] DeadAGI(SIP/100-081e3648,
agi://localhost/ManageCalls.agi?when=after) in new stack
[Jul 8 17:31:14] ERROR[2960]: utils.c:966 ast_carefulwrite: write() returned
error: Connection refused
[Jul 8 17:31:14] WARNING[2960]: res_agi.c:222 launch_netscript: Connect to
'agi://localhost/ManageCalls.agi?when=after' failed: Connection refused
-- Executing [...@pstn2:2] Dial(SIP/100-081e3648,
SIP/lo...@pstn2/011212664800450||S(20)) in new stack
-- Setting call duration limit to 20 seconds.
[Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host:
pstn2/011212664800450
[Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
my extention.conf:
[pstn2]
exten = h,1,DeadAGI(agi://localhost/ManageCalls.agi?when=after)
exten=_!X.,1,GotoIf($[${EXTEN:0:1}=1]?internal:external)
exten =_!X.,n(external),Dial(SIP/lo...@pstn2/011212664800450,,S(20))
my sip.conf
[general]
register=login:p...@host
[pstn2]
type=peer
host=hostname
insecure=invite
nat=yes
qualify=yes
secret=secret
username=username
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
allow=alaw
fromdomain=domaineName
[100]
secret=100
username=100
type=friend
context=pstn2
nat=yes
disallow=all
allow=ulaw
allow=gsm
allow=alaw
host=dynamic
i do not know why it prints No such host: pstn2/011212664800450??
Any suggestion
De : Paul Belanger paul.belan...@polybeacon.com
À : Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Envoyé le : Jeu 8 juillet 2010, 12h 10min 14s
Objet : Re: [asterisk-users] Re : Communication IAX2 SIPIAX2
On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote:
But it does not work.
Any suggestion
Without posting a debug log it makes it hard to troubleshoot.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.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
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http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
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Re: [asterisk-users] [NAT] * + private IP + locked-down firewalls?
On Fri, Jul 9, 2010 at 4:28 AM, Gilles codecompl...@free.fr wrote: On Mon, 05 Jul 2010 12:45:34 +0200, Gilles codecompl...@free.fr wrote: Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP... For those interested, I was tipped through private e-mail about using OpenVPN to open a steady tunnel between the client and Asterisk, and have the SIP client send packets through that tunnel instead of trying to connect directly. I have around 50 Snom 370s configured this way. They work great for remote workers. However the Snom speakerphone is terrible compared to Aastra and Polycom. If there is any background noise it will cut in and out the other party. Ryan -- _ -- 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] Sip Proxy
Can i make build Proxy server by asterisk -- Best Regards M.D -- _ -- 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] Re : Re : Re : Communication IAX2 SIPIAX2
On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote: ok it works i had a problem with a syntax: i had to wrire: exten =_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20)) Correct, Dial(SIP/lo...@pstn2/011212664800450,,S(20)) Is not valid syntax -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
[asterisk-users] Pbx för Windows?
Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! -- _ -- 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] Pbx för Windows?
YATE FreeSWITCH available on windows. Asterisk can be build for windows using cygwin. There are some PBX software also available on windows but with some limitation. Signatures fai...@vopium.com Regards, Faisal Hanif VoIP Manager m +45 72 72 00 01 m +92 32 1405 9996 Vopium A/S | Office No.2, 7th Floor, Shaheen Complex| Abbot-Road, Lahore, PAKISTAN 5400 t +92-42-3631-6491 | f +92-42-3631-6492 | w www.vopium.com Think about the environment before printing this mail P Tnk p miljet fr du printer denne mail On 7/9/2010 5:41 PM, Christian wrote: Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! -- _ -- 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 and cisco 2800
If you do back to back, then one end needs to clock. To set it on the Cisco, type clock source internal under the controller config. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, July 09, 2010 4:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 2800 Hi Peder, it seems to work, thank you! Now I've got a problem with the cisco 2800 which is resetting every 5 minutes but I do not think it is related to the cable, maybe something about the clock but except for a wiki page (http://www.voip-info.org/wiki/view/Asterisk+legacy+integration) there is nothing on internet about connecting asterisk and cisco... :( Giorgio Incantalupo Peder wrote: That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Tuesday, July 06, 2010 2:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 2800 Hi Neeraj, my problem is not terminating but making the Cisco accept the calls coming from my Asterisk. The bad news is I cannot have access to the Cisco sw, it is like a black box for me. The only thing I can have access to is the T1/E1 port on the back of the Cisco 2800. I made a custom cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri, 02 Jul 2010 18:54:31 +0200 From: Giorgio Incantalupo gincantal...@fgasoftware.com Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4c2e19c7.5090...@fgasoftware.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2 15:20:36] VERBOSE[15004] logger.c: -- Accepting AUTHENTICATED call from XXX.XXX.XXX.XXX: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:1] Dial(IAX2/1-1024, DAHDI/g2/X|60|tT) in new stack [Jul 2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Everyone is busy/congested at this time (1:0/0/1) [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Executing [6...@inbound:2] Hangup(IAX2/1-1024, ) in new stack [Jul 2 15:20:36] VERBOSE[15031] logger.c: == Spawn extension (inbound, , 2) exited non-zero on 'IAX2/1-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- 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] Pbx för Windows? - Email found in subject
Mayby Freepbx. http://www.freepbx.org/ Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Christian Verzonden: 09-07-2010 14:41 Aan: asterisk-users@lists.digium.com Onderwerp: [asterisk-users] Pbx för Windows? - Email found in subject Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! -- _ -- 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] Pbx för Windows?
Christian a écrit : Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! Maybe he can try something like this : http://www.3cx.com/ -- Cordialement / Greetings -- _ -- 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] Pbx för Windows?
On Fri, 9 Jul 2010, Christian wrote: Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Er, how about Asterisk? http://www.asteriskwin32.com/ However, there's 3CX: http://www.3cx.com/ip-pbx/asterisk-on-windows.html Hm. They're using intersting SEO techniques to promote 3CX with that search term though... However I have a reseller who's installing 3CX rather than my Asterisk boxes and thy seem to be getting on well with it. Gordon -- _ -- 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] Pbx för Windows?
On Friday 09 Jul 2010, Christian wrote: Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! My best tip is just to install Linux and Asterisk on a separate machine. You don't need a particularly high-spec box to run it; anything over 1 GHz ought to be fine. -- AJS -- _ -- 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] Pbx för Windows?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, July 09, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pbx för Windows? On Friday 09 Jul 2010, Christian wrote: Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! My best tip is just to install Linux and Asterisk on a separate machine. You don't need a particularly high-spec box to run it; anything over 1 GHz ought to be fine. -- AJS -- -- Assuming you don't need DAHDI lines, an easy solution would be to run SwitchVox in a VMplayer session. _ -- 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] Pbx för Windows?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Friday, July 09, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pbx för Windows? On Fri, 9 Jul 2010, Christian wrote: Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Er, how about Asterisk? http://www.asteriskwin32.com/ However, there's 3CX: http://www.3cx.com/ip-pbx/asterisk-on-windows.html Hm. They're using intersting SEO techniques to promote 3CX with that search term though... However I have a reseller who's installing 3CX rather than my Asterisk boxes and thy seem to be getting on well with it. Gordon -- -- The 3CX option might be palatable; if I were going with the asteriskwin32, I would just download cygwin and build a current 1.4 or 1.6 branch (branch used is 1.2.26.2) _ -- 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] Pbx för Windows? - Email found in subject
Arjan Kroon | Mobillion wrote: Mayby Freepbx. http://www.freepbx.org/ And, as their page states, FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Pbx_för_Windows?_-_Email_f ound_ in_subject
I echo the sentiment that you should just run Asterisk on some small hardwarein an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: Re: [asterisk-users] Pbx_för_Windows?_-_Email_found_ in_subject From: Doug Lytle supp...@drdos.info Date: Fri, July 09, 2010 8:17 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Arjan Kroon | Mobillion wrote: Mayby Freepbx. http://www.freepbx.org/ And, as their page states, FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Call failed: 408 timeout
Hello: Here is my sip and extentions configuration and the log of x-lite, because i don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i hope you can help me. SIP.conf [default] include=anexos include=anexos1 include=anexos2 [anexos] exten= 100,1,Dial(SIP/100,0) exten= 100,2,Hangup [anexos1] exten= 101,1,Dial(SIP/101,0) exten= 101,2,Hangup [anexos2] exten= 102,1,Dial(SIP/102,0) exten= 102,2,Hangup EXTENTIONS.CONF bindport=5060 ; bindaddr=0.0.0.0 srvlookup=yes (10:23:26) : [100] type=friend secret= callerid=javier100 host=dynamic disallow=all allow=all context=default nat=no [101] type=friend secret= callerid=informatica101 host=dynamic disallow=all allow=all context=default nat=no [102] type=friend secret= callerid=admin102 host=dynamic disallow=all allow=all context=default nat=no LOG X-LITE (10:16:24) : © 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1105d build stamp 9 License key: 31AC0B511918201B7ED760CE6BC073B6 Established SIP protocol listen on: 10.44.1.20:5060 Firewall Discovery Skipped SIP: 10.44.1.20:5060 RTP: 10.44.1.20:8000 NAT: 10.44.1.20 SEND TIME: 3079422292 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079423989 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079427009 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079433234 SEND 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica sip:1...@10.44.1.20;tag=93961341 To: sip:100 Contact: sip:1...@10.44.1.20:5060 Call-ID: 716c23be-d94f-060d-2f43-d9557f056...@10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 -- _ -- 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] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
Thanks fro the input. The area is a 4 square feet. So, you are saying that if I use four speakers then they would not be as loud as needed? Thanks again 2010/7/9 Massimo Nuvoli mass...@archivio.it bruce bruce ha scritto: Hi Guys, I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2 Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet height. Is that enough? Is there calculator online I can use to determine the number of speakers needed? I guess these speakers go in chain so I am not sure if the full capacity of the speaker (30 watt) will be used. Hu interesting... i never checked this kind of product. CyberData has calculator only for the 8W model, but... every speaker they sell is 8w and the calculator say 69 speakers. You can attach 2 speakers to one amplifier in parallel (they say this also), this is the maximum as the amplifier cannot reach 32W (4 speakers), but you can try to use 4 speakers (2 in parallel + 2 in parallel) with a little less than maximum 8w on each speaker. I think a reasonable number of speakers may be less than half, but you must check wath is in the area, also remember if this is a warehouse to place the speakers where a person can be, not goods. :-) For a so big installation think to use a voip interface and professional product with low voltage line speakers, i think is less expensive. Bye. -- _ -- 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] Call failed: 408 timeout
SEND 0.0.0.100:5060 ?! -- _ -- 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] VoIP Users Conference Recordings
/R, It's guest # 3 on the call 2day. Sorry but where exactly are we on the call? I can't seem to find the website you are demoing. Help! On Sat, Jul 3, 2010 at 3:09 AM, Randy R randulo2...@gmail.com wrote: Hi, Alistair Cunningham of Integrics was our guest yesterday. We talked about Integrics new product Geons, a suite of software for building large-scale distributed enterprise applications. The recorded session is now available here: http://www.voipusersconference.org/2010/geons/ The extremely rare John Todd was sighted (and heard) at this event. If you are developing a product or service involving VoIP we would love to hear from you. Contact me off list or via the VUC site if you or someone from your project or company would like to be a guest or with suggestions for future guests and/or topics. Next week our guest is Quickfuse (http://quickfuseapps.com) a product that allows you to quickly build IVR by drawing it on a page. You can take it for a test run free and then join us to ask questions or give feedback on Friday July 9th at 12 Noon EDT. Small world: Finally, as a result of being a VUC regular, Michael Iedema of Askozia embedded pbx met Randal Schwartz at Astricon last year and this week will be a guest on FLOSS Weekly July 7th. http://twit.tv/FLOSS /r -- _ -- 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] Re : Re : Re : Communication IAX2 SIPIAX2
Glad you found the issue, sorry for not being able to help. 2010/7/9 Paul Belanger paul.belan...@polybeacon.com On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote: ok it works i had a problem with a syntax: i had to wrire: exten =_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20)) Correct, Dial(SIP/lo...@pstn2/011212664800450,,S(20)) Is not valid syntax -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- Abdullah -- _ -- 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] Call failed: 408 timeout
Hi, Please disable firewall and SElinux. 2010/7/9 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de SEND 0.0.0.100:5060 ?! -- _ -- 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] power outage
I have a TE205P that has been working fine for 2 years. power outage yesterday took out my everything for over an hour. Everything has come back up except the PRI. My provider has checked it to the box and says everything looks good on their end. I get this message: [Jul 9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! ztcfg -vvv Zaptel Version: 1.4.12.1 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 7 channels to configure. and show status gives me condition RED of course. How do I find out whats wrong here? Jerry -- _ -- 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 Crashes - Segmentation Fault
Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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] Delay between answer and pickup ?
We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing hello ? Hello ? and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of asterisk don't show this discrepancy. We are using a local channel to dial a landline through a sip provider. When the call is answered, the agent's phone is then dialled. the logs go something like this [Jul 9 13:29:26] VERBOSE[23396] logger.c: [Jul 9 13:29:26] -- SIP/provider-0001ed6e is making progress passing it to Local/somenum...@dialleroutbound-4c93,2 [Jul 9 13:29:44] VERBOSE[23396] logger.c: [Jul 9 13:29:44] -- SIP/provider-0001ed6e answered Local/01577864...@dialleroutbound-4c93,2 .. [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Executing [*00...@diallerconnected:2] Dial(Local/somenum...@dialleroutbound-4c93,1, SIP/*0086*|5|iA(autoanswer)) in new stack [Jul 9 13:29:45] VERBOSE[23416] logger.c: [Jul 9 13:29:45] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- Local/somenum...@dialleroutbound-4c93,1 requested special control 20, passing it to SIP/*0086*-0001ed73 [Jul 9 13:29:46] VERBOSE[23416] logger.c: [Jul 9 13:29:46] -- SIP/*0086*-0001ed73 answered Local/somenum...@dialleroutbound-4c93,1 .. as you can see, the call is answered at 13:29:44 and the agent gets called (auto-answer phones) at 13:29:46, yes if you listen to the call recording, there is a 6 second gap between the person saying hello and the agent being connected. Is it possible that the call was answered 5 seconds *before* I get notification of the answer ? i.e. is the provider taking too long notifying me of the answer ? Julian -- _ -- 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] no subject
Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike -- _ -- 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] False answer() being sent by cellphone providers
On 7/9/10 9:57 AM, Mike Ely mike...@amyskitchen.net wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike Argh, got distracted, here's the version with a Subject: header. -- _ -- 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] False answer() being sent by cellphone providers
On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] chan_iax2: I should never be called!
Hi, Recently, one of my Asterisk servers stopped connecting calls and required a reboot to fix it (did not try to restart or reload). The log showed loads of this message: NOTICE[302] chan_iax2.c: I should never be called! This highly repeated message seems to be preceded by something like: WARNING[10767] channel.c: Exceptionally long voice queue length queuing to IAX2/coinbound-15879 When this happens it also seems that SIP peers on a gigabit LAN start going on/offline frequently. So that seems to explain why calls start to fail. There is absolutely nothing wrong with the network (and switches). I don't know if it can be a NIC problem on the server but how can I tell? [Jul 9 08:10:49] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2819ms / 2000ms) [Jul 9 08:10:50] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (860ms / 2000ms) [Jul 9 08:10:51] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2003ms / 2000ms) [Jul 9 08:10:52] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (876ms / 2000ms) [Jul 9 08:10:54] NOTICE[10756] chan_sip.c: Peer '7054' is now Lagged. (2929ms / 2000ms) [Jul 9 08:10:56] NOTICE[10756] chan_sip.c: Peer '7054' is now Reachable. (963ms / 2000ms) [Jul 9 08:11:03] NOTICE[10756] chan_sip.c: Peer '7054' is now UNREACHABLE! Last qualify: 3096 Rebooting the server solved everything... for now... Any ideas? Vieri -- _ -- 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] power outage
Jerry Geis wrote: and show status gives me condition RED of course. What's the output of pri show span 1? Check your cable. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] False answer() being sent by cellphone providers
Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- _ -- 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] False answer() being sent by cellphone providers
On 7/9/10 10:29 AM, Mike Ely mike...@amyskitchen.net wrote: Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? Sorry, bad joke. In all seriousness though, is there not a way to detect this behavior and handle the answer() correctly? -- _ -- 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] General network question regarding SIP and IAX2
Hi all, i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network? What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the routing? From my understanding SIP s used to find the way to the remote party and the voice is transferred over RTP directly from client to client without permanently involving the Server. IAX seems to do all in one, the routing and the transport of the voice. Is that correct? Why i am asking this? Lets say i have one Asterisk running in London and another one in Paris. Both are connected via IAX2 trunk over a WAN connection. User A is registered on the server in London. User B is registered on the server in Paris. Now User A is visiting User B in Paris and both have call with each other. Is the voice data routed from user A to Asterisk in London and then back via IAX2 to the server in Paris and the to user B? Or is there a direct connection between them and no WAN traffic is produced? And is there a difference between using either SIP or IAX as client protocol in that case? I hope i explained well what i meant. Thanks in advance for answers. -- _ -- 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] no subject
On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote: Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? *CLI core show application AMD -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
Re: [asterisk-users] False answer() being sent by cellphone providers
On 07/09/2010 12:33 PM, Mike Ely wrote: On 7/9/10 10:29 AM, Mike Ely mike...@amyskitchen.net wrote: Some of the systems blokes might just figure that¹s another collections agent and hang up then ;) On 7/9/10 10:09 AM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? Sorry, bad joke. In all seriousness though, is there not a way to detect this behavior and handle the answer() correctly? The Dial() application can already play an announcement to the called party and wait for them to confirm the call before accepting that the outbound channel is 'answered'. This allows your dialplan to go on to another party to call if the first does not actually accept the call. This is useful both in the case you describe, and when the outbound call gets delivered to voicemail, since that appears to be 'answered' at the network level as well. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] power outage
On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis ge...@pagestation.com wrote: and show status gives me condition RED of course. Physical problem, check cables / telco -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
Re: [asterisk-users] General network question regarding SIP and IAX2
The variable is *canreinvite.* *Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etcif you are two phones are closer to each other than a round trip to Asterisk server.* * * *On the flip side, you can't record these calls because no media is sent through Asterisk.* * * *-Bruce * On Fri, Jul 9, 2010 at 1:48 PM, unsero...@aol.com wrote: Hi all, i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network? What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the routing? From my understanding SIP s used to find the way to the remote party and the voice is transferred over RTP directly from client to client without permanently involving the Server. IAX seems to do all in one, the routing and the transport of the voice. Is that correct? Why i am asking this? Lets say i have one Asterisk running in London and another one in Paris. Both are connected via IAX2 trunk over a WAN connection. User A is registered on the server in London. User B is registered on the server in Paris. Now User A is visiting User B in Paris and both have call with each other. Is the voice data routed from user A to Asterisk in London and then back via IAX2 to the server in Paris and the to user B? Or is there a direct connection between them and no WAN traffic is produced? And is there a difference between using either SIP or IAX as client protocol in that case? I hope i explained well what i meant. Thanks in advance for answers. -- _ -- 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] False answer() being sent by cellphone providers
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, July 09, 2010 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- It is a DAHDI function that you may or may not get a reliable notification of answer. The best thing to do is to MOH for 7 seconds, then play a message this is a message from the computer room; press 1 to accept. This lets you not waste time on a not real answer. Here is a cliff-note context: [accept] exten = s,1,Answer exten = s,n,WaitExten(7) exten = s,n,Background(important) exten = s,n,WaitExten(5,m) exten = 1,1,backgrounf(message) exten = 1,n,hangup exten = t,1,hangup exten = i,1,hangup exten = *,1,hangup -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] False answer() being sent by cellphone providers
Do some R D with asterisk function AMD (Answering Machine Detection) if that can help you. Regards, Faisal Hanif On 7/9/2010 11:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, July 09, 2010 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- It is a DAHDI function that you may or may not get a reliable notification of answer. The best thing to do is to MOH for 7 seconds, then play a message this is a message from the computer room; press 1 to accept. This lets you not waste time on a not real answer. Here is a cliff-note context: [accept] exten = s,1,Answer exten = s,n,WaitExten(7) exten = s,n,Background(important) exten = s,n,WaitExten(5,m) exten = 1,1,backgrounf(message) exten = 1,n,hangup exten = t,1,hangup exten = i,1,hangup exten = *,1,hangup -- _ -- 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] General network question regarding SIP and IAX2
Sounds great, thanks for your answer. Do i need to set this on the trunk, the friend or on both? -Original Message- From: bruce bruce bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, Jul 9, 2010 8:13 pm Subject: Re: [asterisk-users] General network question regarding SIP and IAX2 The variable is canreinvite. Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etcif you are two phones are closer to each other than a round trip to Asterisk server. On the flip side, you can't record these calls because no media is sent through Asterisk. -Bruce On Fri, Jul 9, 2010 at 1:48 PM, unsero...@aol.com wrote: Hi all, i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network? What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the routing? From my understanding SIP s used to find the way to the remote party and the voice is transferred over RTP directly from client to client without permanently involving the Server. IAX seems to do all in one, the routing and the transport of the voice. Is that correct? Why i am asking this? Lets say i have one Asterisk running in London and another one in Paris. Both are connected via IAX2 trunk over a WAN connection. User A is registered on the server in London. User B is registered on the server in Paris. Now User A is visiting User B in Paris and both have call with each other. Is the voice data routed from user A to Asterisk in London and then back via IAX2 to the server in Paris and the to user B? Or is there a direct connection between them and no WAN traffic is produced? And is there a difference between using either SIP or IAX as client protocol in that case? I hope i explained well what i meant. Thanks in advance for answers. -- _ -- 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 -- _ -- 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] [Dahdi 2.3.0.1] Does it need all those modules?
Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1. compiled and installed Dahdi 2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist netjet and unblacklist wctdm: == # cat /etc/modprobe.d/dahdi.blacklist.conf blacklist wct4xxp blacklist wcte12xp blacklist wct1xxp blacklist wcte11xp blacklist wctdm24xxp blacklist wcfxo #blacklist wctdm blacklist wctc4xxp blacklist wcb4xxp blacklist netjet == 3. rebooted, and checked that netjet was gone and wctdm was in: == # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? Thank you. -- _ -- 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] [NAT] * + private IP + locked-down firewalls?
On Fri, 9 Jul 2010 08:06:04 -0400, Ryan Wagoner rswago...@gmail.com wrote: I have around 50 Snom 370s configured this way. They work great for remote workers. However the Snom speakerphone is terrible compared to Aastra and Polycom. If there is any background noise it will cut in and out the other party. Thanks for the feedback. It's good to know that there's an almost-guaranteed solution to the one-way audio. -- _ -- 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] [Dahdi 2.3.0.1] Does it need all those modules?
I just went through a Dahdi rebuild, and I seem to recall a message that all modules will be loaded until you set up the dahdi configuration files. regards Scott On 7/9/2010 11:41 AM, Gilles wrote: Hello To use Dahdi + Asterisk with a PCI card with a single FXO port, I just... 1. compiled and installed Dahdi 2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist netjet and unblacklist wctdm: == # cat /etc/modprobe.d/dahdi.blacklist.conf blacklist wct4xxp blacklist wcte12xp blacklist wct1xxp blacklist wcte11xp blacklist wctdm24xxp blacklist wcfxo #blacklist wctdm blacklist wctc4xxp blacklist wcb4xxp blacklist netjet == 3. rebooted, and checked that netjet was gone and wctdm was in: == # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? Thank you. -- _ -- 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] [Dahdi 2.3.0.1] Does it need all those modules?
On 07/09/2010 01:41 PM, Gilles wrote: 3. rebooted, and checked that netjet was gone and wctdm was in: == # lsmod | grep -i wc wctc4xxp 32414 0 dahdi_transcode 5751 1 wctc4xxp wcb4xxp33905 0 wcfxo 8968 0 wctdm24xxp116684 0 wcte11xp 22995 0 wct1xxp12971 0 wcte12xp 26308 0 dahdi_voicebus 39947 2 wctdm24xxp,wcte12xp wct4xxp 230713 0 wctdm 35677 0 dahdi 197809 11 xpp,dahdi_transcode,wcb4xxp,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp,wctdm crc_ccitt 1339 3 wctdm24xxp,dahdi,hisax == Does Dahdi really need all those modules, or is there another configuration file that I missed to disable unneeded modules? /etc/dahdi/modules controls which modules /etc/init.d/dahdi will load on start. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] False answer() being sent by cellphone providers
Thanks for the tip! On 7/9/10 11:35 AM, Faisal Hanif fai...@vopium.com wrote: Do some R D with asterisk function AMD (Answering Machine Detection) if that can help you. Signatures fai...@vopium.com Regards, Faisal Hanif On 7/9/2010 11:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, July 09, 2010 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. How about a loop with Please press pound to continue? -- It is a DAHDI function that you may or may not get a reliable notification of answer. The best thing to do is to MOH for 7 seconds, then play a message this is a message from the computer room; press 1 to accept. This lets you not waste time on a not real answer. Here is a cliff-note context: [accept] exten = s,1,Answer exten = s,n,WaitExten(7) exten = s,n,Background(important) exten = s,n,WaitExten(5,m) exten = 1,1,backgrounf(message) exten = 1,n,hangup exten = t,1,hangup exten = i,1,hangup exten = *,1,hangup -- _ -- 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] General network question regarding SIP and IAX2
I guess it has to be on the Trunk and one of the either user or peer and the opposing party shouldn't have it as no. But, to full proof urself, put it on the trunk and both users. Basically put it anywhere that takes it. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite-Bruce On Fri, Jul 9, 2010 at 2:40 PM, unsero...@aol.com wrote: Sounds great, thanks for your answer. Do i need to set this on the trunk, the friend or on both? -Original Message- From: bruce bruce bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, Jul 9, 2010 8:13 pm Subject: Re: [asterisk-users] General network question regarding SIP and IAX2 The variable is *canreinvite.* *Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etcif you are two phones are closer to each other than a round trip to Asterisk server.* * * *On the flip side, you can't record these calls because no media is sent through Asterisk.* * * *-Bruce * On Fri, Jul 9, 2010 at 1:48 PM, unsero...@aol.com wrote: Hi all, i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network? What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the routing? From my understanding SIP s used to find the way to the remote party and the voice is transferred over RTP directly from client to client without permanently involving the Server. IAX seems to do all in one, the routing and the transport of the voice. Is that correct? Why i am asking this? Lets say i have one Asterisk running in London and another one in Paris. Both are connected via IAX2 trunk over a WAN connection. User A is registered on the server in London. User B is registered on the server in Paris. Now User A is visiting User B in Paris and both have call with each other. Is the voice data routed from user A to Asterisk in London and then back via IAX2 to the server in Paris and the to user B? Or is there a direct connection between them and no WAN traffic is produced? And is there a difference between using either SIP or IAX as client protocol in that case? I hope i explained well what i meant. Thanks in advance for answers. -- _ -- 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 -- _ -- 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] Logging codec used in CDR
Happy Friday everyone, Is there a way to log the negotiated codec that was used for each call in CDR or in a separate log file? This is for SIP-based calls, if that matters. Perhaps there is some variable that can be queried as part of the dialing script; Or is it possible to grab the codec name using the exten =h, after the call completes... Thanks in advance for all suggestions. S -- _ -- 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] Problem with call-limit
I have the same problem, I have asterisk 1.4.21.2. I have limitonpeer = yes in context general, call-limit=10 in all peers, but still have this message in Cli. 2010/7/8 Jonas Kellens jonas.kell...@telenet.be Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies DB I have a column call-limit which has a value of '4' for all the sip peers. Still I get the above message... 2nd situation : I should be possible to transfer a call by pressing # followed by the extension, but it does not work. Although I have a call-limit of '4' and thus the peer I'm transfering to should be able to receive the transfer. [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin '#' received on SIP/test13-000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF begin passthrough '#' on SIP/test13-000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end '#' received on SIP/test13-000b, duration 320 ms [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end accepted with begin '#' on SIP/test13-000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end passthrough '#' on SIP/test13-000b [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- Started music on hold, class 'default', on SIP/test3-0007 [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- SIP/test13-000b Playing 'pbx-transfer' (language 'be') [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '2' received on SIP/test13-000b [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '2' on SIP/test13-000b [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end '2' received on SIP/test13-000b, duration 320 ms [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF end passthrough '2' on SIP/test13-000b [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '0' received on SIP/test13-000b [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin ignored '0' on SIP/test13-000b [Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end '0' received on SIP/test13-000b, duration 320 ms [Jul 8 09:46:58] DTMF[22334] channel.c: DTMF end passthrough '0' on SIP/test13-000b [Jul 8 09:47:01] VERBOSE[22334] logger.c: [Jul 8 09:47:01] -- Stopped music on hold on SIP/test3-0007 [Jul 8 09:47:01] -- Executing [...@from-test:14] Dial(SIP/test3-0007, SIP/test2) in new stack [Jul 8 09:47:01] WARNING[22334]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Jul 8 09:47:01] == Everyone is busy/congested at this time (1:0/0/1) Anyone know the problem with call-limit ?? Kind regards, Jonas. -- _ -- 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] Logging codec used in CDR
Hi! Is there a way to log the negotiated codec that was used for each call in CDR or in a separate log file? Use CHANNEL(audionativeformat) - and do the same with the help of the M option to Dial() for the remote call leg. Store that info in the CDR userfield, or create your own field if you are on Asterisk 1.6 with the adaptive CDR columns. Note: In principle the codec can be changed in the middle of the call, however in practice this very rarely (never) happens. And while you are at it also look at RTCP stats as well. If you are still on Asterisk 1.4 then consider to apply bug/patch #10590. More details: http://www.voip-info.org/wiki/index.php?page=Asterisk+RTCP Philipp -- _ -- 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] Pbx_för_Windows?_-_Email_f ound_ in_subject
Hi all, Many thanks for your replies! Will tell my friend and see what he will be interested in. Many thanks! Christian -Ursprungligt meddelande- Från: mgra...@mstvp.com Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Skickat: 10-07-09 15:29 Ämne: Re: [asterisk-users] Pbx_för_Windows?_-_Email_f ound_ in_subject I echo the sentiment that you should just run Asterisk on some small hardwarein an appliance like fashion. In fact, just yesterday I posted an overview of hardware suitable for DIY appliances. I've used many of the platforms mentioned. http://www.mjgraves.com/2010/07/08/d-i-y-asterisk-appliances-a-question-of-scale/ Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves Original Message Subject: Re: [asterisk-users] Pbx_för_Windows?_-_Email_found_ in_subject From: Doug Lytle supp...@drdos.info Date: Fri, July 09, 2010 8:17 am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Arjan Kroon | Mobillion wrote: Mayby Freepbx. http://www.freepbx.org/ And, as their page states, FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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.co -- _ -- 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] False answer() being sent by cellphone providers
On Fri, 9 Jul 2010, Mike Ely wrote: On 7/9/10 9:57 AM, Mike Ely mike...@amyskitchen.net wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike Wow. So presumably you start to pay for the call before the mobile phone actually rings and you answer the mobile phone? So you're charged even if the mobile phone user doesn't answer? Are you sure? Although I guess it's a country specific thing - if they tried that over here I think it'd be pitchforks and flaming torches at their UK HQ offices... Gordon-- _ -- 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] False answer() being sent by cellphone providers
On 7/9/10 3:20 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Fri, 9 Jul 2010, Mike Ely wrote: On 7/9/10 9:57 AM, Mike Ely mike...@amyskitchen.net wrote: Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, Mike Wow. So presumably you start to pay for the call before the mobile phone actually rings and you answer the mobile phone? So you're charged even if the mobile phone user doesn't answer? Are you sure? Although I guess it's a country specific thing - if they tried that over here I think it'd be pitchforks and flaming torches at their UK HQ offices... Gordon (off list) Yes indeed we do. The telcos here are absolutely abhorrent, to the point that much could be written about how horrible they are but nobody would want to read such depressing material. And consumer protections? Hah! The devotees of Ayn Rand have written most consumer law here. Don't get me started. -- _ -- 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] False answer() being sent by cellphone providers
On Fri, 9 Jul 2010, Mike Ely wrote: (off list) Continuing to veer off-topic... Yes indeed we do. The telcos here are absolutely abhorrent, to the point that much could be written about how horrible they are but nobody would want to read such depressing material. And consumer protections? Hah! The devotees of Ayn Rand have written most consumer law here. Don't get me started. Maybe you should re-read Atlas Shrugged. Your laws may have been written by the people Ayn Rand wrote about: the government increasingly asserts control over all industry, while society's most productive citizens, led by the mysterious John Galt, progressively disappear,* not devotees of Ayn Rand and her philosophy. *) http://en.wikipedia.org/wiki/Atlas_Shrugged -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] False answer() being sent by cellphone providers
-Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards Sent: Fri 7/9/2010 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject:Re: [asterisk-users] False answer() being sent by cellphone providers On Fri, 9 Jul 2010, Mike Ely wrote: (off list) Continuing to veer off-topic... Yes indeed we do. The telcos here are absolutely abhorrent, to the point that much could be written about how horrible they are but nobody would want to read such depressing material. And consumer protections? Hah! The devotees of Ayn Rand have written most consumer law here. Don't get me started. Maybe you should re-read Atlas Shrugged. Your laws may have been written by the people Ayn Rand wrote about: the government increasingly asserts control over all industry, while society's most productive citizens, led by the mysterious John Galt, progressively disappear,* not devotees of Ayn Rand and her philosophy. *) http://en.wikipedia.org/wiki/Atlas_Shrugged Well, so much for my off list attempt. Perhaps I should learn how to use email before I take on anything so complex as a PBX. At any rate, Steve, you have it completely backwards: in the US and many other countries, it is industry asserting control over government, not the other way around. Walk down K Street in Washington, D.C. and you'll see my point. And no thanks: I've already read that execrable book, and found it to be nothing more than overwrought claptrap written to give people with a huge inferiority complex (witness all the carping on about mediocrity) some smug self-justification when they abandon all ethics in favor of their reptile-brain, base instincts. Disgusting. Cheers! -- _ -- 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] PHP can't insert - Can someone please help
Hi Guys,
I am making another module for Voicemail. I have three fields in a POST form
that have to be connected together to make it a single 10 digit number but
there is something wrong in my syntax probably.
$npaa = ('$_POST[anpa]');
$nxxa = ('$_POST[anxx]');
$blocka = ('$_POST[ablock]');
*$grplist = $npaa.$nxxa.$blocka;*
$sql=INSERT INTO findmefollow(grpnum, strategy, grptime, grppre, grplist,
annmsg_id, postdest, dring, needsconf, remotealert_id, toolate_id, ringing,
pre_ring)
VALUES
('$_POST[grpnum]','ringall','$_POST[grptime]','$_POST[grppre]',$grplist,'0','$_POST[postdest]','','','0','0','Ring','$_POST[pre_ring]');
It seems that $grplist is the problem. Can someone please point what is
wrong?
Error:
Error: You have an error in your SQL syntax; check the manual that
corresponds to your MySQL server version for the right syntax to use near
'('333')(''),'0','ext-local,vmb2000,1','','','0','0','Ring','0')' at
line 3
Thanks,
Bruce
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Re: [asterisk-users] False answer() being sent by cellphone providers
From: asterisk-users-boun...@lists.digium.com on behalf of Steve Edwards Continuing to veer off-topic... Maybe you should re-read Atlas Shrugged. On Fri, 9 Jul 2010, Mike Ely wrote: And no thanks: I've already read that execrable book, and found it to be nothing more than overwrought claptrap written to give people with a huge inferiority complex (witness all the carping on about mediocrity) some smug self-justification when they abandon all ethics in favor of their reptile-brain, base instincts. Disgusting. Ouch. I think one of my favorite ox has been gored. Maybe we both got what we wanted from the book. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 Crashes - Segmentation Fault
Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) On Fri, Jul 9, 2010 at 8:39 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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
