Re: [asterisk-users] What is the best way to campaign dial 5000 numbers? Spool files or AMI actions?
On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs wrote: > Hi everyone, > > Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about > 5000 numbers and then put the call to agents right away and pull up the CRM > based on the number dialed. So, I am going to be doing some PHP+Ajax work. I > am familiar with spool files but I don't like the fact that I can't read the > status of the call in real-time. However, I know that it's the easiest way > to approach the issue. The way to call 5000 numbers is to call one number, really well. Then you put it in a loop. You need to run a lab for long enough that you have the bugs worked out, before you subject real people to problems. With asterisk you can always tell the real-time status of a call, even if you initiate from a call file. Perhaps you would enjoy reading up on Local channels. Some people prefer to initiate calls from AMI. I tried it and didn't like it. But because most of us have been annoyed by an autodialer in our lives, even if we ourselves have made autodialers in the past, this is probably about the limit of the help you're going to get, unless you ask a more specific question that shows you've been trying to learn this hands-on and you've gotten stuck on a particular problem. -- _ -- 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] Executing Script after MixMonitor is called
On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards wrote: > The OP was using MIXMONITOR_EXEC (although I wonder about the '&&' syntax) > so he doesn't need to explicitly execute (via system()) his commands. Wow. Never knew that was possible. I still don't like the syntax, but good to know. For optimal debugging, I would suggest OP not try to build the pyramids with a single line of commands joined by ampersands, but rather do a separate System() call for either doing my script idea or doing the individual commands in multiple separate System() calls. -- _ -- 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] Executing Script after MixMonitor is called
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg wrote:
> shebang /path/to/bash
>
> PATH=$1
> lame --arguments $1.wav $1.mp3
> if [ -f {$1}.mp3 ] ; then
> rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug. (gotta close the if with
a fi, I declare PATH and never use it, and I'm not certain that the
declare will work without a 'declare' statement)
Your dialplan sample was missing an essential step, which was
System()
that actually kicks off your command. You could also use it to kick
off the bash script, like
System(path_to_script path_argument_for_script)
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Re: [asterisk-users] Executing Script after MixMonitor is called
On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib wrote:
> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor command is
> called but it doesnt execute the command, I tried the command in terminal it
> worked, any help please ... below is my dial plan
> exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F
> -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
> "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f
> "/var/spool/asterisk/monitor/${CALLFILENAME}.wav")
> exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)
One obvious thing to try is to make sure that when you're making your
shell test that you are running the shell as the same user that
asterisk is running as.
Set your console to at least verbose = 3, and run the dialplan. What
do you get for the error?
I will say just looking at that code, you have a mess. It would be
simpler to debug if you made a short shell script that consisted of
something like
shebang /path/to/bash
PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
rm {$1}.wav
and then call that script with the argument that is the path to the
file. Will make your dialplan easier to debug, and still get you the
equivalent functionality.
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Re: [asterisk-users] Odd DTMF problem when receiving calls
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie wrote: > I've been up and down this issue for a few hours and I cannot for the > life of me determine why simply defining a peer causes Asterisk to offer > telephone-event. I have tried specifying dtmfmode=rfc2833 or > dtmfmode=auto in [global] and neither change has any effect. As I said > above, I've copied every configuration directive in [global] into the > peer definition for the company PBX, and calls still work. > > So I'm at a loss to explain this. The problem does not seem to stem > from my configuration, but I'm not entirely sure what else could be the > problem... an Asterisk bug perhaps? I don't want to jump to that > conclusion since this is my first day tinkering with the software. > Perhaps someone more knowledgeable can steer me in the right direction? Wheee. You don't say anything about what 'company PBX' is, so we just have to guess based on your description. Based on your description, your 'company PBX' requires that the endpoints it communicates be registered before-hand. Having a definition for the sip peer in asterisk makes asterisk continually register with the 'company PBX'. So it is not necessarily your case that you think it is, that rfc2833 is required, but rather that for 'company PBX', any sip endpoint must first be registered. Both asterisk and 'company PBX' probably support large numbers of possible DTMF or signaling possibilities, and it's not surprising that you can get away with several possible values. If you do 'sip show peers', with and without the config in your sip.conf (use ; to comment it out and 'sip reload' to commit your changes), you should be able to verify that THIS is the real problem. And no, this is not an asterisk bug. -- _ -- 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-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
On Wed, Jan 4, 2012 at 4:45 PM, Asterisk Development Team wrote: >The Asterisk Development Team is pleased to announce the first >release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0. >2.6.0 is a feature release which: > > wct4xxp: Expose serial number in dahdi_device and kernel log. This is VERY exciting news. Thank you very much! So much easier than having to schedule time to open a box when you forgot to write down the serial number when doing the rack-and-stack. -- _ -- 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] Anyone have a reliable T.38 Solution
On Thu, Jan 5, 2012 at 8:05 AM, Steve Underwood wrote: > No PAP2 or PAP2T supports T.38, even though many people will swear that they > do. For a little while there was some beta code for the PAP2T with badly > broken T.38 support. Perhaps this is where the "legend of T.38 on a PAP2T" > started. Of course, on the internet, when someone posts an incorrect message > many people would like to believe is right, a 1000 people cite it as proven > fact. Thanks for clearing that up. I was getting all excited that I could flash the PAP2T; I've always used regular voice tones over SIP with the PAP2Ts. -- _ -- 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] 1.6 and 1.8
On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas wrote: > Can somebody point me to an explanation from Kevin or Tzafir or someone else > "up the food chain" explaining the differences/benefits of 1.6/1.8 vs > 1.4/10.0? What's the difference between a car released in 2006 versus a car released in 2012? Very complex question. Aside from reading the changelogs, like others have suggested, you should understand that asterisk is developed in a very public way, with open access to source, bugs, and the ability of lots of people to provide patches. Some feature requests are very difficult to honor without a major rewrite of various layers. And the intervening releases made something substantially better (well, some people will disagree; if you don't need any of the newer features, you might be able to ram more calls through an older version) or added more flexibility or more features. If you are running 1.4 and want a feature that comes built-in with 1.6, nobody is going to have any sympathy for you when you say you wish 1.4 did X feature. Basically, if you want to participate in the public ecosystem where you can find something you want improved, or you come across a bug, you need to be at the stage of development where code changes are still happening. -- _ -- 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] Using asterisk with DSP chips
On Fri, Nov 18, 2011 at 2:23 PM, Sazzad wrote: > Hi, > I have to use asterisk with some dedicated DSP chips, which will do the > expensive G729 CODEC computing, so that the server processor has minimum > load. I was informed, I've to use GPAK to implement this. So far I've I had never heard of GPAK before your post, so I had to look it up. I still do not really understand it. There are at least a few ways to combine asterisk and G.729. And honestly, I would say your problem is not so much the cpu load, but the licensing, because G.729 is under patent. You can buy G.729 licenses from Digium, and do G.729 natively in asterisk. http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC > Whether embedded scenario is a must for such development or is it possible > to use asterisk on server and dsp's through some PCI/Ethernet? It is also possible to have asterisk speak open codecs, like G.711, and have another ethernet-connected device to the codec, and pay a different vendor for that licensing. For example, you can use a Cisco 3945, load it up with DSPs, and setup the router config to do the transcoding between G.711 and G.729. I've done this before, and it works, but you have to shell out $10k + for the router and smartnet and all the DSPS if you are doing as many simultaneous channels as I'm using. But I'm still not sure this is your actual question. I realize English is probably not your first language, however, if you contracted for a job and you HAVE to use GPAC because somebody else told you that is the job requirement, I have no idea how to do that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available
On Thu, Nov 10, 2011 at 12:24 PM, Leif Madsen wrote: > On 11-11-10 12:12 PM, Danny Nicholas wrote: >> Yeah! My boss will be much happier having a system that doesn't have the >> -tail on it. > > I hear this kind of statement every once in a while, which makes absolutely no > sense to me. If you're blindly running a version of any software in production Was thinking the same thing. And that it would be nice if all it took to make my boss happy was backspace on a tarball filename. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording a meetme conference
On Thu, Oct 27, 2011 at 11:53 AM, Mike wrote:
> I am trying to record a MeetMe conference, and this is what is relevant in
> the 1.8 manual:
>
>
>
> r - Record conference (records as MEETME_RECORDINGFILE using format
> MEETME_RECORDINGFORMAT. Default filename is
> meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav.
>
>
>
> Which is fine, but , but how do I change the default filename. Let’s say I
> want /tmp/recording.wav?
There is not a rule against using Monitor() or MixMonitor() on the
entire channel. This might give you the control (format, filename) you
are looking for.
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Re: [asterisk-users] Running as non-root
On Sun, Oct 23, 2011 at 3:16 PM, Tzafrir Cohen wrote: > On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote: > >> If you use DAHDI, you need to change ownership of /dev/dahdi/* to the >> non-root owner. I ended up rolling that into the init script for >> dahdi. > > The init script of DAHDI or asterisk is the wrong place for that. > > If you're one of those who actually uses static files, you set their > permissions at creation time or whenever. > > The rest of you: set the permissions in udev rules, as in the ones > included with DAHDI. This avoids any potential races and unnecessary > work. Thanks for the tip. I just noticed that the permissions 'came undone' if I did a DAHDI reload, so it seemed like the right place. For the record, I'm also using SNMP with asterisk, also as non-root, and I'm also having a problem with /var/lib/masterx or whatever also reverting to being owned by root. And again, my presumptive fix is to put the chown directly into the SNMP script. Ideas? -- _ -- 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] Running as non-root
On Wed, Oct 19, 2011 at 7:19 AM, Torbjörn Abrahamsson wrote: > Thank you, I actually found the asterisk.conf settings after sending the > mail. So next question is which folders/files do I need to change ownership > of to make it work? > > > > /etc/asterisk > > /var/lib/asterisk > > /usr/lib/asterisk > > /var/spool/asterisk > > /var/log/asterisk > > > > And the files in them of course… Any more? Any that should not be in the > list above? If you use DAHDI, you need to change ownership of /dev/dahdi/* to the non-root owner. I ended up rolling that into the init script for dahdi. -- _ -- 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 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen wrote: > However you could select/deselect modules using menuselect if you wanted to > automate the process. It's documented over here: > > http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439 Yeah, menuselect has been in linux kernel tree forever, and I was excited when it got added to asterisk build. But that was irrelevant because I wasn't getting the configure to complete. The configure is a prerequisite for any of the make commands. I'm also a big fan of 'make sounds'. -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming wrote: > This is a bug in the configure script, but in the meantime, you should be > able to use "--without-pwlib" to avoid it, as long as you aren't trying to > build chan_h323. Thanks much. I was trying ./configure --disable-chan_ooh323 and that was not making a difference. -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... /usr/bin/ptlib-config ./configure: line 24978: 2*1+6*100+5 /usr/lib64/ /usr/lib/: division by 0 (error token is "/lib64/ /usr/lib/") There seems to also be a problem with CentOS 6 in general that I have not found a package that actually provides /usr/bin/ptlib-config. I copied that binary over from a CentOS 5 install to see if I could get my original error to clear. Here's THAT error... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... no ./configure: line 24906: --ldflags: command not found Cannot find ptlib-config - please install and try again And then I searched around the config trying to figure out how not to use ptlib at all, and ultimately I tried just doing... rpm -e ptlib ptlib-devel (and the other packages) And this got my asterisk configure to complete. But really, I'm wondering if other people have run into the same problem, or if there is a nifty configure argument that would keep me from needing to uninstall the library. -- _ -- 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] upgrading from 1.4.39 to 1.8.5
read the 1.6 README and the 1.8 README. If you're using SIP you should expect changes with account authentication, faxing, output regarding channel status and performance. I think that version of 1.4 is late enough you would already be on DAHDI for hardware devices. If not, you need to convert to DAHDI. There are also features you might want to try out. Nobody but you knows how you use your system. Do it on a test system before you put it into production. On Fri, Sep 2, 2011 at 8:16 PM, Joseph wrote: > What sort of things should I watch out for when upgrading from 1.4.39 to > 1.8.5 > > Thanks, > -- > 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 > -- _ -- 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] Faxes suddenly failing
That debug looks cool but I have no idea what it means. If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor. When you can hear the audio of the fax hopefully you will be able to tell what's going on, and if you're lucky it's something specific to the particular kind of testing you are doing. I don't think it's an exaggeration to say there have been hundreds of posts over the last few years about broken T.38. Avoid it in favor of traditional audio faxing. Even if you can control both endpoints, there's just so much that can go wrong when faxing over voip. If you need this to be 'reliable faxing', you should seriously consider doing your faxes over copper. If you cannot afford that, it should be a top-tier voip provider on a dedicated line, where you will not be starving for bandwidth, and you should never compress the audio on those calls. On Wed, Aug 31, 2011 at 7:27 PM, C F wrote: > I think you should change the subject line to: > Faxes suddenly worked for 2 weeks. > > On Wed, Aug 31, 2011 at 3:49 PM, Tim King wrote: >> I realize that faxing is not great with voip but here is my confusion. I >> have been working on a web based fax system for 2 weeks. During this time I >> have sent over 100 2 page faxes without any errors. Now today as things are >> finally completed I can not seem to get any fax to go through unless it is a >> 1 page cover only. Anyone able to tell the issue from this debug output? >> >> -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started >> -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE >> rt: IDLENSRX >> -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY >> rt: RRDYNHRY >> -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED >> -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI >> -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.091837 ], stack sent 5 frames (100 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.160248 ], stack sent 3 frames (60 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.960201 ], channel sent 48 frames (960 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 000.979464 ], channel sent 1 frames (20 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 003.157848 ], stack sent 150 frames (3000 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 003.219814 ], stack sent 3 frames (60 ms) of silence. >> -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP >> rt: WDSRNT21 >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 005.579811 ], stack sent 118 frames (2360 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 006.481179 ], channel sent 275 frames (5500 ms) of silence. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 007.801045 ], channel sent 66 frames (1320 ms) of energy. >> -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP >> -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP >> rt: NT4X >> -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP >> rt: UNEXPECT >> -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END st: WT_DIS_RSP >> rt: RXXXNFRX >> -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI >> -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 011.152812 ], stack sent 279 frames (5580 ms) of silence. >> -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP >> rt: WDSRNT21 >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 013.471827 ], stack sent 116 frames (2320 ms) of energy. >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 014.260642 ], channel sent 323 frames (6460 ms) of silence. >> -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 016.460661 ], channel sent 110 frames (2200 ms) of energy. >> -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS >> -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP >> rt: WDSRNDCS >> -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 >> -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH >> -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 >> -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 >> -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM >> -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM st: WT_DIS_RSP >> rt: WDSRNSWE >> > Channel 'SIP/MyVoipProvider-0046' fax session '12', [ >> 016.540315 ], channel sent 4 frames (80 ms) of silence. >> -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN >> rt: UNE
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards wrote: > Many times, I've made the statement that you can execute hundreds of AGIs > written in C in the time it takes to load an interpreter and parse a script > written in PHP or Perl. I've truly enjoyed this thread. And while startup time is certainly part of the equation, I'm curious whether you also recorded memory overhead while you were doing your benchmarks. In my personal observation of 'Perl running in production', there's a significant memory complexity associated with running lots of simultaneous Perl interpreters. I'm guessing the PHP overhead is smaller but still a factor. Because C lib is both statically compiled and essentially built into the system, there's no such 'waste' of running the interpreters. Along with the narrowing gap you saw by 'throwing hardware at the problem' that shrinks the difference between C and interpreted languages, I personally now work with a production asterisk environment where systems have dozens of GB of ram. As such, it's not entirely crazy to say 'so what' about the whole thing. Nonetheless, do you have any numbers to back up my theory? -- _ -- 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] ReceiveFax to G.711
On Mon, Jun 27, 2011 at 9:06 AM, Michael wrote: > Hi Kevin, > > Controlling it through the sip.conf peers is sufficient for us for this case > (because this particular provider doesn't support T.38 at all), but I think > it would be a good idea to add the option to enable/disable T.38 from the > dialplan. If I recall correctly, that's how callweaver worked at the time. > > Also, we just checked it, and since for that provider, we have other codecs > in higher priorities (like GSM, for example) than G.711, G.711 was not > chosen as the only codec, so the fax transmission failed. We can not > prioritize G.711 over the other codecs in the sip.conf, for the obvious > reasons, so for this, we need to do it in the dialplan. How can we do it? Obvious solution is to move this particular number off this particular provider to a provider where you don't require prioritizing GSM and where that provider properly supports T.38. Outside of that idea, you would have to do some edge SIP routing before the call hit asterisk, to deflect this particular call to an asterisk configured differently. You could do that with OpenSIPS for example. Doesn't that sound like a lot more work than just porting the number to a different provider? Or if you insist on keeping the provider, get a second account with them for just this number, and you can apply different rules in sip.conf for just that account. -- _ -- 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] t.38 virtual fax software?
On Fri, Jun 24, 2011 at 4:55 PM, Hose wrote: > Can anyone recommend some kind of virtual t.38 fax software? I'd like > to test/debug some of the t.38 stuff, but it'd be much easier if I had a > software client that could just generate the faxes from a workstation, > rather than having to sit with the fax machine + t.38 ata to source > faxes from. another machine or virt + asterisk + callfiles that invoke SendFax() > There doesn't seem to be much out there, and the stuff that's out there > is kind of expensive for me just to be using for testing. That's because callfiles are free -- _ -- 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] UK English sounds packs
On Thu, May 26, 2011 at 9:09 AM, Ishfaq Malik wrote: > Hi > > Does anyone know if there are any free UK accented English sounds packs? Which UK accent? I'd pay a small bit of money just to hear a Liverpool accent set of voices. You can do some Scottish voices with Festival and the "Alan" voice. They do sound robovoice though. http://www.cstr.ed.ac.uk/projects/festival/onlinedemo.html -- _ -- 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] receive faxes
On Thu, May 5, 2011 at 1:43 PM, vip killa wrote: > The majority of open source projects out are NOT run by commercial > institutions... Postfix kicks butt. But only because IBM paid for development, for a long number of years, and because they hired somebody who had a really good idea how to improve Sendmail. Asterisk kicks butt. It just does, even if you can't get your silly faxing edge case to work, because Digium paid for development, for a long number of years. There are also people who come along, and because the source is available, fix a particular bug that annoyed them, or added a feature they personally wanted. T.38 has a boatload of problems, and most of those problems are because people who aren't employed by Digium did not read the specs, or they did read the specs, but felt like they had to violate the specs to get their code to work with a different broken T.38 stack. I've personally fixed problems with asterisk, and I found my code contribution accepted. Perhaps you've submitted patches the bug tracker? We eagerly anticipate your voluntary code contributions. Maybe statistically, there are more open source projects out there that have non-paid lead developers, and they do their work on their own dime, and on their own time. But unless you're lucky enough to live in a place where you can hunt and gather for everything you need, you need money to live in a society, and have the resources to be able to sit at a keyboard long enough to churn out code. I could only afford to make my 'code contribution' because I have a day job where I needed the fix. -- _ -- 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] receive faxes
On Wed, May 4, 2011 at 12:00 PM, A J Stiles wrote: > (For my part, I'm actually surprised that nobody came up with a proper > protocol for encapsulating the stream of zeros and ones that make up a fax > transmission but rely on the precise timing inherent with a circuit-switched > network, into something more suitable for sending over a packet-switched > network. That would have fixed it good and proper.) They did. It's called TCP / IP. It allows sending PDFs, and they can even be encrypted. Faxing is for people who haven't heard of the internet. -- _ -- 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] The new ConfBridge application is now in Asterisk Trunk!
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich wrote: > > > Does this ConfBridge requires a hardware timing source? No, and neither does MeetMe with modern DAHDI. > Will I be able to use this on any virtual server without having the need > special changes to > the VM setup? Define 'any'? If you're idea of virtualization is to oversubscribe servers and hurt performance, then no. To mix audio, the code takes lots of audio slices and merges them with an algorithm. But if the underlying cpu doesn't provide consistent, reliable ticks, as potentially happens in virtualization, then good luck with what's going to happen to your audio mixing. -- _ -- 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] The new ConfBridge application is now in Asterisk Trunk!
On Mon, Apr 25, 2011 at 9:38 AM, David Vossel wrote: > I am proud to announce that after a good bit of development, community > feedback, testing, and >code review, the brand new ConfBridge application has > been officially merged into Asterisk >Trunk!!! > http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 > > If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, > forget everything you >know. This is a completely revamped, highly > optimized, and feature rich conferencing >application capable of mixing > sample rates from 8khz all the way up to 192khz! Exciting right?! So way back when the 'old' ConfBridge was announced, my understanding was it was originally an internal Digium tool for exercising the Bridge() code and it was decided to release it to the public in the event the code might be useful to others. The old ConfBridge was missing stuff that was in MeetMe(), and wasn't that compelling for my particular usage. This 'new' ConfBridge looks to be much more full-featured. So can anybody explain the motivation for this? Is this a replacement for MeetMe() where at a certain point we envision dropping MeetMe() from the codebase? Does ConfBridge() scale to many users as nicely as MeetMe? I'm assuming the MeetMe ability to use a hardware source for timing will still be superior with large user counts in rooms? -- _ -- 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] Fax
On Fri, Apr 1, 2011 at 7:04 AM, Khaled W. Chehab wrote: > 1-Is there a way to export fax tiff file image from .pcap captured file . Maybe, but I can't think of how. If you can somehow invert the pcap file back into packets and reproduce the fax traffic, then maybe. > In other words i am trying to backup all faxes that are passing on my > network,and export the fax file later on. > Is this feasible and how . I don't know why you wouldn't want the tiff immediately. Traditionally people use fax because it's faster than putting a letter in the mail. I don't know why anybody would want slower fax. All I can come up with is you're trying to evade toll charges or otherwise change one expensive call into two cheaper calls. If that's what you want to do, it would be even cheaper to skip the fax, and use the internet to exchange a pdf file between the actual two endpoints. As to whether it's possible? Maybe if instead of the pcap, you do analog fax, record the call, and play back the recording. Maybe that would work? > 2-I tried using asterisk to receive the fax and resend it , but that will > not a real time fax, and if I want >to ignore the time ,in case of failed > fax I should inform the user that fax is been failed since the system > >received the fax is the first place successfully . If you put a fax machine in the way of another fax machine, it's not going to be real-time. Any kind of arrangement involving two faxes like that is going to be store-and-forward. > Is there a way to don’t hangup the call when the user send the fax first > time to system ? Yes, but you cannot stop the sending party from hanging up their leg of the call. -- _ -- 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] T.38 negotiation error
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.) : > Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. > When I try to send a fax in T.38 mode I receive this error > > ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel > 'SIP/eutelia-sirio-out-' is in an unsupported T.38 negotiation > state, cannot continue. > > In my sip.config general section I have added this lines > t38pt_udptl=yes > t38pt_rtp=no > t38pt_tcp=no > > If I comment this lines, the fax is sended by G711 but this modality is too > slow and instable. It depends on who you're faxing with. If you can control who you'll be T38-ing with, you might be able to solve this particular problem. But if you're T.38-ing with the whole world, in my opinion, there are too many broken T.38 implementations for you to depend on this. If you want reliable faxing, you should avoid voip altogether, and use a modem. I've gotten pretty good faxing by using audio passthrough, over sip, but it's only sip on a LAN, to a PRI gateway system. Trying to do voip fax over internet is cheap, but it's not reliable. -- _ -- 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] ReceiveFAX issue.
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman wrote: > Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't > make t.38 work I keep getting the following error "Disconnected after > permitted retries" Any ideas on this? So you're saying if you turn off t38 in sip.conf, you receive faxes successfully? Problem solved. Don't use T.38 in your particular environment. -- _ -- 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] ReceiveFAX issue.
On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman wrote: > Do you know how to force off T.38 in res_fax? it's in sip.conf take a look for t38pt_udptl=yes change it to no > reload sip on your console that should force it to either fail entirely or do audio passthrough. -- _ -- 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] ReceiveFAX issue.
On Tue, Jan 25, 2011 at 9:34 AM, Bryant Zimmerman wrote: > On 01/24/2011 2:54PM Bryant Zimmerman wrote > The attached file was too large so I am putting in a link to the file. It is > a virus free text file. You failed to mention earlier that this is T.38. Turn off T.38 and see if it's still broken. T.38 is notorious for having broken implementations by multiple vendors. And upgrading to latest asterisk may or may not improve the situation. -- _ -- 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] U-verse DTMF tuning for Zaptel
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards wrote: > We know the problem exists -- the boss just installed U-verse at his house > :) > It works fine from cell and copper, just not from U-verse and their ilk. Well, I would say more data samples are needed then. It could certainly be the boss's uverse connection and not other uverse connections. If every SIP connection failed wouldn't you know it by now? -- _ -- 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] U-verse DTMF tuning for Zaptel
On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards wrote: > One of my clients is complaining that their customers that use U-verse (and > other cable providers) for telephone service cannot enter credit card > numbers reliably. > > The issue not all digits are received in my dialplan. > > The calls come in on PRI. > > It's an old 1.2 install, so the only tweak available is 'relaxdtmf.' > > Any clues on how to proceed? > > Would jumping to 1.6 help? Can you record a few calls just to confirm the problem? I've had 'dtmf issues' that really were cases of people entering data during recordings rather than at the appropriate time in the prompts. -- _ -- 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] ReceiveFAX issue.
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman wrote: > I am testing out inbound faxing using res_fax and res_fax_spandsp.so > > My system answers the call but then sets there on the ReseiveFax line then > comes back with an error that it exceeded the maximum retries. > How would I go about debugging this? Below is my very simple dialplan code I > am using, and the fax show version gives the following as well. Record the call with Monitor() or MixMonitor(). Listen to the call. See if you can figure out anything obvious. Why are you doing a Wait(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
Re: [asterisk-users] Asterisk to asterisk t.38
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal wrote: > I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can > send recieve faxes from both boxes fine to and from pstn. But the faxing > between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- 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] ReceiveFax
On Thu, Jan 20, 2011 at 10:00 AM, Flavio Miranda wrote: > Hi all, > I realize that the application Receivefax can't handle with more than one > fax at the same time. In a environment with a lot of fax, some caller get > the signal but the operation can't be completed. > Is there a way to send busy tone to the second caller? This would be a problem with your underlying channel, not something intrinsic to ReceiveFax. If you only have a single copper pots line, then you would need to get more physical lines, or find another way to get more circuits into your PBX. Rest assured that nothing about ReceiveFax stops you from answering multiple calls at once, provided you have the underlying capacity. -- _ -- 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] Weird phone behavior after recent CentOS 5 update
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham wrote: > For some reason our Asterisk box is doing something really unusual following > applying a routine update to CentOS 5 on Monday. > > We have Asterisk 1.4.2 and its been working great for years. But now when > the phone system receives an incoming SIP call, its not providing any audible > dial sound to any caller. It is recognizing the incoming call, and after no > answer for about 5 rings or so, it goes to voice mail. But there is no > audible 'ring' to the caller. Just nothing - blank, empty silence. > > Of course any automated answering system (ie. business phone menu, etc.) that > we have works just fine. Its just the lines that go directly to an internal > phone that are no longer providing any audible ring which is sending a > message to the caller that their call didn't go through. > > Does anyone have any idea what might cause this? Definitely check your firewall settings. Definitely consider rebuilding against the libs as they now exist on your machine. CentOS (which are really RedHat) library changes aren't always fully disclosing the things that actually change. The worst example in recent memory was when a CentOS update changed the defaults to sudo, and you had to go manually override to allow sudo to work without a tty. -- _ -- 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] cdr_mysql stopped working
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman wrote: > I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql > table for CDR's today there are no entries since the update. > I have rebuilt and re-installed and re-started asterisk still no CDR's > flowing to mysql. I did not change any configs. I checked to make sure that > the cdr_mysql option was selected under the make menu options. The module > shows it is there when I do a modules show. I don't get any errors saying it > can't write to the table. My voicemail settings are pulling from the same > server. > > Any ideas on what I could try to fix this or how I could test to see what is > causing it? Rebooting is a good clue. You could check your firewall settings. Firewalls can stop mysql connections. Try manually connecting to the mysql server from the asterisk system and see what happens. -- _ -- 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] [POTS/BRI] Neutral comparisons of PCI vs. box?
On Wed, Dec 8, 2010 at 10:17 AM, Gilles wrote: > On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg > wrote: >>* pay somebody else to do it in the form of appliance and lose most >>control versus do it yourself and have total control but also the >>chance to screw up. > > Thanks for the input. Has someone in this ng tried a PCI card and then > an appliance of the same grade, and could provide some feedback? I've done both. It's hard to determine what 'same grade' means. I guess you could go by price? I've used Digium PCI cards. They worked well. I've used Cisco gear for terminating PRI lines. It also worked well. Both the cards and the appliances have had 'issues'. The Digium PCI cards themselves were solid, but there are lots of ways to do the configuration wrong. You have to match the line settings to the line provided by the phone company, but sometimes you figure those out by trial and error. That's essentially no different for the appliances. With asterisk there are certain bugs I've run into, but unlike with Cisco, I've been able to find the actual problem, change the code, and run the version I want, which also has the patch I want. I've gotten into an upgrade / downgrade trap with Cisco where I wanted to fix a PRI error so I upgraded. Then the upgrade generated broken callerID. I couldn't downgrade if I wanted the PRI fix, and I wanted the PRI fix more than I wanted callerID. Alas, Cisco. All known, documented issues, escalated through Cisco paid support. -- _ -- 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] [POTS/BRI] Neutral comparisons of PCI vs. box?
On Wed, Dec 8, 2010 at 9:06 AM, Gilles wrote: > Hello > > I need to find a recent and neutral comparison of the major products > available to connect an Asterisk server to the telephone network, > whether ISDN (BRI) or PSTN, and through a PCI card or some external > box. I'm told there are less issues (echo, stability) with external > boxes compared to PCI cards. > > Apparently, the main brands are Digium, Sangoma, Rhino Equipment, > Patton, and Audiocodes. I would disagree that there are fewer issues. I would phrase it as: * pay somebody else to do it in the form of appliance and lose most control versus do it yourself and have total control but also the chance to screw up. These are the choices you face in much of tech, such as develop software yourself that does exactly what you want versus buy somebody else's software that does some of what you want. If you want to pay somebody else to do it, there's also the middle road of pay somebody who knows how to do PCI cards to do it for you. We call those people consultants or vendors. You can try hiring the vendors directly or ask them for recommendations for vendors trusted to do a good job. -- _ -- 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 with MySQL Cluster
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson wrote: > I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant > High-Availability. I was wondering if it's possible for Asterisk to also > use multiple database servers for Realtime? Currently with Realtime I am > only able to point to a single IP address for a database. If that database > server goes down that Asterisk is pointed to then Asterisk won't be able to > do anything. Any options within Asterisk 1.8 to make it more fault tolerant > when it comes to Realtime and databases? http://dev.mysql.com/doc/refman/5.0/en/ha-overview.html It's a fair amount of work for what in my opinion is a minimal reward. If you've hardened everything else and this is the only single point of failure left in your entire infrastructure, you should be able to sleep well at night. -- _ -- 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 Log viewer
On Tue, Nov 23, 2010 at 8:25 AM, voip crazy wrote: > Hello, > > I want to analyze the asterisk logs files, looking for all kind of > errors, ¿Anyboby knows any asterisk logs analyzer? You're only going to have the logs for what you create logs for. I create custom logs for the custom things I need answers on, like IVR button presses, user actions, authentication, etc. -- _ -- 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] Stability..
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub wrote: > Sorry, > what I meant was: > server*CLI> remove extension (hit tab) > segfault.. > 1.4.22 > It could be an extension name Where is the error trapping if this is the > case.. Who writes this shit? If you remove an extension that is being used, control could flow into the now non-loaded extension, and THAT is what caused your core dump. Don't do that. If you want to NEVER load extensions and also not crash asterisk, you're better off taking a look at modules.conf, particular making entries that begin with noload => You can also take a look at 1.6 and make menuconfig, and just not build modules that you don't want. Then they'll really never load at startup. -- _ -- 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] Timing cable usage necessity
2010/11/25 Захаров Антон : > Hello everyone. > > I have a timing slips errors and I can't understand what source of the > problem is. > My installation has 2 digium cards: TE420 and TE220 cards in one server. > There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - > normal installation for transit communication. > Span configuration is: > span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. > span=2,0,0,ccs,hdb3 #TE420 - second port. To PBX. > span=3,2,0,ccs,hdb3 #TE420 - third port. To PSTN. > span=4,0,0,ccs,hdb3 #TE420 - fourth port. To PBX. > span=5,3,0,ccs,hdb3 #TE220 - first port. To PSTN. > span=6,0,0,ccs,hdb3 #TE220 - second port. To PBX. > I should to say, that PBXs are interconnected through router (doesn't > know anything about it). So all schema looks like this: > http://yfrog.com/jjschemaj > Spans 1-5 works fine, but on span 6 (marked bold) I have rising timing > slips counter. > > I think it's appearing because I'm getting a primary timing source on > span 1 - first port on TE420. But TE220 doesn't use it's span 5 for > timing source, because it has priority 3, so it could be a sync problem. > Am I wrong? > > I'm started to think about timing cable for syncing timing on first card > and second. Should I use it? > > It's a problem to bought cable in our city (Russia,Moscow). All > resellers sell only cards. Could I use floppy or IDE cable to > interconnect cards? As I see in picture of cable, it's a direct 16 pin > cable. > > Does anybody know something about timing cable for different cards? How > I can solve my problem? I can't answer your problem about how you find one in Moscow, but I can tell you that when I've installed two cards in a single server I have used the timing ribbon cable. I have no idea whether it made a difference, and I now build my servers differently. You could ask Digium directly, as they probably know their resellers. Something that may help you is that you can make sure you set the priority on the cards differently from each other. There is a dial with a pointer that tells you which card is prioritized. Whenever I had two in the same server I made sure that one card was set higher than the other. -- _ -- 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 recording format
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius wrote: > Hi All, > We have a requirement to record over 60 simultaneous calls. Our recording > facilities are implemented using Monitor() over AMI. The thing we have > noticed that making 60 simultaneous call recordings using wav CPU load is > significantly higher (around 2 times more) than using gsm. Even writing call > recordings to /dev/null makes a big difference in CPU load. Ignoring your real questions, and asking an alternate question: Why not just record in gsm? If your answer is that you have to play these back on Windows, you can build an on-the-fly gsm-to-wav converter using sox. My understanding is that recording in wav doesn't exactly make you have higher audio quality in your recordings, although the experts at codecs could better answer that. -- _ -- 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] Volume on meetme recording
On Mon, Nov 15, 2010 at 8:30 AM, Richard Kenner wrote: > It's kind of low for me. How does one control that volume? I've never heard of a way to control that volume. You can tweak after-the-fact with sox, or you can crank up your soundcard / amplification on playback. -- _ -- 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] Big practical systems
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch wrote: > But can anyone contribute some practical knowledge of systems that take in > channel bank T1s or DS3s from "far away", and process the calls? Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a channelized DS3 into PRIs. > Not to start the discussion, but "Is there a board that will take a DS3 (672 > channels) and a system that will handle the calls, or is that a silly > question?" If by board, you mean PCI board for shoving in something with an intel cpu, not that I've ever heard. Digium sells 4x port PRI boards, and some competitor sells an 8x port PRI board, but I've never tried any boards not made by Digium. The only thing silly is the idea of trusting that many calls to PC hardware. > Is there an IP box that would take the DS3 and then a system that would > handle the calls? Yes, embedded hardware from a vendor you've heard of will do that. Cisco makes a 3845 which can terminate about 20 PRIs in one appliance. > My guess would be yes because the actual call load would > be far lower than 672 calls. Maybe 100-150 or so simultaneous. Well, then it's not really a DS3. If it can't do the whole thing without melting down, it shouldn't advertise itself as DS3. The Adtran gear works rock solid when pushed to the limit. If you're just talking 150 calls, you could do that with two 4x port cards in a single PC. I thought you were talking a lot bigger. > Each line/call would have to have absolute caller ID. In other words, PSTN > call handling. Ummm, there's no such thing as absolute caller ID. You wanna try that question again? callerID is not legally binding, is not used by billing, anybody can spoof it. The closest you can get is to have a LEC provide ANI. You don't need PRI to get that. You can get that via a quality voip provider, or yourself using your own termination gear to convert into voip. -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question wrote: > Another question: Is there (expect for the admin guide that we didn't > succeed to understand the example in) an example somewhere for ReceiveFax > full extensions.conf diaplan? We would like to allocate one of the > extensions that our SIP provider gives us to a fax storage server or later > to email. Not that I've ever seen. I built mine by reading mailing list archives, then the source for app_fax. +1 for open source. At least one reason such a thing does not exist is that everybody has a different idea of what 'full extensions.conf dialplan' means. In my case, I ReceiveFax, record the call, give it a naming convention that works, convert tiff to pdf, tail a log to a flat txt file, copy the pdf to Winders. That's not going to be the same thing as what a lot of other people want to do. -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question wrote: > Digium claims that their FFA is the best and most compatible solution and > they give one channel for free, but do not provide support for those that do > not buy more channels, but why buy more channels if the free/test one > doesn't work? I don't know. I'm not using FFA, and I'm doing more channels simultaneously than I want to disclose. It's called app_fax, and it's been built into the 1.6 series for quite some time now. You seem pretty hell bent against spending any money. Not for FFA, not for commercial digium support, not for an analog copper line, not for an ATA. So try app_fax. It's free. -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question wrote: > The whole point (as I specified in the header and initial message) is the > attempt to use "Fax for Asterisk" to send the message. Asterisk can handle audio passthrough faxing. I'm talking audio faxing over SIP. You compile against this thing called SpanDSP, and then asterisk squawks audio tones over the line. It's amazing. Until you've tried it, you don't know whether it could work. I'm under the assumption that you'd rather have faxing at all than faxing over T.38. The world is littered with broken T.38 implementations. Just because it's a standard, doesn't mean people follow it. Ever heard of HTML? Which browsers follow it to the letter? You seem to have never successfully exchanged a fax with your target, so I don't know why you think the far end isn't broken. Try turning off t38 and see what happens. -- _ -- 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] integrate Intertel Axxess with Asterisk
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst wrote: > How did the setup work as far as extensions on the Inter-Tel system > contacting extensions on the asterisk system? It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel worked. Still had to watch out for line error counters, and still had to reboot it daily (Windows + Inter-Tel equals unstable). When sending calls into Inter-Tel, the other side, probably asterisk, masquerades as telco sending in call as PRI. In the other direction, you configure the lines as OPX, or off-premise extensions. Just make the extensions match for each line on the PRI and set your dialplan so you keep things making sense. -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question wrote: > It's set to yes for this peer. > > also t38pt_udptl is set to yes. > > :( You don't say anything about what you're trying to send / receive against. Here's how you should troubleshoot: * start with a 'real fax machine' if you have one, on an analog line if you have one. If you can't receive / send with that against your target, blame your target. * move to audio-pass through fax on asterisk. No T.38. If that works. * add in T.38 You will learn things in that process and be able to tell at what layer your troubles are happening. It could be coincidental that things give up during the reinvite. It could actually be giving up for noise on the line, packet drops, etc. At the very least, start recording the call. You'll at least be able to hear up to the re-invite. Definitely record the audio passthrough attempt and listen back to it. -- _ -- 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] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question wrote: > Hello, > > I'm trying to send a tif file, using Fax for Asterisk and the call is > executed, but when I get the reINVITE with T.38 data, the local server > doesn't recognize that we have this capability and sends a 488 message. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite take a look at your canreinvite option. -- _ -- 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 insert dynamic hostname into shared CDR database
So I'm in a situation where I want to consolidate cdr logs. My general
idea is to use cdr_mysql for this.
I know I can do things like
Set(CDR(userfield)=hostname)
And I can hardcode the hostname for the dialplan on each system.
But what I'd really like to do is have this dynamic, so I can use the
same exact dialplan on multiple hosts, and have hostname dynamically
generate.
I went looking for a built-in asterisk variable for this, but haven't
found anything yet.
I know I can do something like
Set(HOSTNAME=(System(/bin/hostname -s))
Set(CDR(userfield)=${HOSTNAME})
But that seems like more overhead than it should take to do this.
Can anybody think of a better way?
Hey, I just discovered
${SYSTEMNAME} * value of the systemname option of asterisk.conf
which was turned off in my asterisk.conf, but then there's an option called
autosystemname = yes
With that enabled, ${SYSTEMNAME} is giving me what I want.
I also spotted
${ENV(HOSTNAME)} which I assume would let me have whatever is set as
an environmental variable. This didn't work for me, perhaps because my
asterisk daemon is running in an environment without that variable
set?
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Re: [asterisk-users] user number in conference
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll wrote:
> Hey,
> i forgot to ask, how can i get the user number from a caller he is in a
> conference, i don't find a variable to us this for the current channel.
> Only the command "meetme list " shows the usernumber, but i can't use
> this output.
why not?
asterisk -rx 'meetme list '
Depending on your version, 1.6 has the concise argument, which
transforms the output into convenient exclamation-point-separated
output.
Then you can send it off to awk -F'!' and pick off the first value.
asterisk -rx 'meetme list concise' | awk -F '!' '{print $1}'
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Re: [asterisk-users] integrate Intertel Axxess with Asterisk
On Wed, Oct 6, 2010 at 5:00 PM, marvin horst wrote: > Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone > system via a SIP trunk using the IPRC card? I have, believe it or not, integrated Asterisk with Inter-Tel. However, not via SIP. Run the costs. When I did, it was way cheaper to integrate asterisk with Inter-Tel via PRI card than via SIP, especially when you figured price per channel. I had a bunch of PRI cards on Inter-Tel talking to asterisk. That was revision one. Revision two, as we got bigger, I went to Cisco gear, like the 3845, and plugged the PRIs from Inter-Tel into the Cisco gear, and used the Cisco gear for the SIP conversion. This let asterisk talk straight SIP and not worry about talking directly to the Inter-Tel. We grew the Inter-Tel to as big as we could get it, offloaded as much as we could, and eventually we couldn't fit our call center into it anymore. Now we're Cisco for the call center. I don't know whether what I used was called Inter-Tel Axxess. I always just called it Inter-Tel. -- _ -- 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] Unable to load fax modules
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati wrote: > thanks for replies, > I am using Asterisk 1.6.2.11 > and components res_fax-1.4_1.2.1-x86_64 and > res_fax_digium-1.4_1.2.1-barcelona_64. > (amd 64 bit machine) > actually I am not aware that there is version which include fax. > for rebuilding with manager support that would be great if you could give me > a link to know about that, cause i've never done it ! If you're building from source, do a make menuconfig then you get the ability to do select/deselct on the individual applications. One is called app_fax That contains the built-in fax support. It requires that you have already pre-built SpanDSP. And also, it looks like you cannot NOT compile in manager support, so my original idea seems wrong. -- _ -- 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] Unable to load fax modules
On Thu, Sep 30, 2010 at 10:51 AM, khalid touati wrote: > Hi List, > I did follow the procedure to install Free Fax for Asterisk successfully > till i came accross this isssue: i can't load the fax module: > > pbx3*CLI> module load res_fax_digium.so > Unable to load module res_fax_digium.so > Command 'module load res_fax_digium.so' failed. > [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error > loading module 'res_fax_digium.so': > /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: manager_event > [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module > 'res_fax_digium.so' could not be loaded. > > any help will be much appreciated!! Don't know for certain, but my guess is you didn't build in AMI or manager support with your asterisk build. Rebuild with manager support and you should be able to find the undefined symbol. And if you're building from scratch, I don't know the reason why you wouldn't use the built-in app_fax.so -- _ -- 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] Record() Cmd and My SQL
On Sun, Sep 26, 2010 at 10:49 PM, Govind, Mahesh (NSN - IN/Bangalore) wrote: > Another reason for storing in the database is to , enable some other > apps to access the recording at some point of time . Yeah, still not a good reason. File systems allow multiple read streams to the same file. Perhaps you should consider a webservice with a squid proxy in front of it if you want to have hundreds of simultaneous accesses. Just store a recording as a file, and if you have metadata, store the metadata in a database. -- _ -- 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] Record() Cmd and My SQL
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly wrote: > Don sez: I don't know how to make Outlook indent. I usually top-post, but I > don't like getting yelled at. > > Why do you say "Don't do that"? Is there a real reason that it would be bad? Performance is a real reason. Multiple simultaneous write streams into a database sounds like a disaster. While trying to read from the db and use it to listen to recordings sounds like a bigger disaster. /path/to/the/recording is a short varchar string the actual recording is a massive, usually multi-megabyte, potentially multi-gigabyte blob. http://en.wikipedia.org/wiki/Blob_(computing) If you're not actually taking advantage of the recording being in the database, doing computing that is easier because of the database, such as nearest neighbor searches, indexing, and the like, you're just slowing down your ability to store and retrieve recordings. > I'd like to put the recordings in a database so they are available to > another application that has no other relationship to the Asterisk server. Sounds like a filesystem. I can store my pdf file with my web browser, and read it on another computer after I store it to my shared filesystem. > The application uses the database to determine if the recording has been > listened to, by whom and if it needs additional attention. Database can maintain metadata (as can a filesystem, owner, creation date, access date), but you could still just store a pointer to the actual file in the db. If you were paranoid about the filesystem and db getting out of step you could do referential integrity checks in the application. If you want to do something wholesale to all the recordings, like carve off the first five seconds, it's quite straightforward with a batched sox call against the filesystem. If you want to do that in a db, it's a select, write output to a file, convert the file, and replace on the value to store it back into the db. -- _ -- 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] Record() Cmd and My SQL
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore) wrote: > The reason is when doing a load balancing , We cannot confine the > recording to a particular asterisk machine ( If we have more than one > asterisk machine in the topology ). Yes you can. You can record the file wherever the call takes place. In fact, you can make the recording on any network segment the packet traverses as well. > So a centralized mechanism might be better . So that any machine can > access the recording . > Regards > Mahesh Recordings are formatted data, typically stored as files. You can put them into a database, but you haven't provided a reason why that would be a good idea. There are these things called shared filesystems. You should take a look at them. They work well. Options include NFS, iscsi, sans, etc. Or you can record the file in-place, and when the recording completes, copy it off to your shared filesystem. That's what I do. Or you can take a look at something like OrecX, which let's you do network spanning on your entire subnet, and it doesn't matter where your call takes place because all RTP streams get written to disk. None of what you've explained would be a good reason to put your recordings into a database. Don't do that. -- _ -- 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] Record() Cmd and My SQL
On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore) wrote: > HI , > > Is there Any way is there so that I can store my recordings directly to a > database rather storing the same to a file . Please, please, please tell us why you would want to do that. -- _ -- 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 T38
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett wrote: > In the simplest terms I can think of, I'm going to describe what I want to > do and I want to know if it's possible in the current version of asterisk. > > Can I take a T38 call from an ATA, convert that back to analog and have > asterisk screech that out on a POTS line to a remote fax machine. Would it > work? > > And could I receive an incoming fax the same way? I suppose that merely removing ATA and asterisk from the middle, and plugging a pots line into a fax machine is out of the question. Sounds like you want a T.38 gateway. Not built into asterisk, but some people have tried patching. Search the archives for T.38 gateway. -- _ -- 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 different is implementing Cisco based system than Asterisk based system?
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote: > Now I have no previous experience with Cisco systems and don't want to screw > up anything. Are they much different than Asterisk based systems? sometimes. Cisco supports "SIP", but depending on the product, asterisk inter-networking with call transfers / dials / etc. can be, ummm, interesting, and you have to do Transfer() rather than Dial() if you want subsequent transfers / conferencing, etc. to work within Cisco. Basically, call setup / control and RTP aren't necessarily on the same device(s) which is the opposite of my asterisk experience. > I guess the underlying VoIP technology is the same for both the systems so it > shouldn't be hard to set it up on Cisco. sometimes. Again, the size of the deployment is relevant here. > Any ideas, suggestions. I'd appreciate your help as what to look for, where > to start from. Again, at some point you'll need to call a reseller. In the meantime, if you want to keep them honest, you should get your hands on paper or digital copies of the Cisco press books about their phone system products. Can't recommend anything specific without knowing things about size and purpose of the install. -- _ -- 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 different is implementing Cisco based system than Asterisk based system?
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote: > Now I have no previous experience with Cisco systems and don't want to screw > up anything. Are they much different than Asterisk based systems? I guess > the underlying VoIP technology is the same for both the systems so it > shouldn't be hard to set it up on Cisco. Any ideas, suggestions. I'd > appreciate your help as what to look for, where to start from. My experience > with Cisco is limited to their networking equipment, IOS, their 7960 series > phones and making them work with asterisk, and also using Cisco press's > wonderful book 'Taking charge of Your VoIP Project'. You don't mention anything about size or nature of the installation. If these people want Cisco support, they'll have to buy this gear from an official Cisco reseller, and that's when you'll get a ballpark price for initial deployment, ongoing support, etc. When you compare that to Asterisk, perhaps you'll have a convert. Or maybe not. Decision makers have their reasons and clearly if price was the only reason, Cisco / Avaya / etc. wouldn't still be selling phone systems. -- _ -- 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 1.6 and fax
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei wrote: > Hello! > I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. > When i try to receive fax I get: > [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel > 'SIP/crocus-ua-0004' refused to negotiate T.38 > [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error > transmitting fax. result=49: The call dropped prematurely. > [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error > > I definitely know that this peer supports T.38 because it works on Lynksys > PAP2T. There are lots of devices that 'support' T.38, but the problem is that they 'support' it differently. If you want to have fun, read the release notes for a Cisco voice IOS, and grep for the word T.38 to see the long list of known broken situations. Just because it's 'supported', doesn't mean it works. Internet Explorer 'supports' html, but good luck getting it to act like a standards-compliant web browser. Try turning off the T.38 and do analog passthrough, or try using two T.38 PAP2Ts. Or even better, don't use fax if you can avoid it. -- _ -- 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 1.6 and fax
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei wrote: > Can you recommend any specific solution to this problem or way to install > app_fax? Not without specific debugging about what problems you're seeing. You get a lot of information when faxes succeed or fail. Try a fax and paste in the debug. -- _ -- 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] Faxes
On Fri, Sep 3, 2010 at 11:50 AM, dave george wrote: > The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk > SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the > PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten => s,1,Playback(silence/1) exten => s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- 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 + cisco 3825 with ISDN
On Tue, Aug 24, 2010 at 9:05 AM, Ron wrote: > hi all, > > i recently subscribe for an isdn and terminate it on a 3825 router. > > i used it as a sip trunk for my asterisk. i'm a newbie when it comes to > ISDN. and i've been experiencing some issues: > > 1. Call Hangup: > > When hangup is initiated from the outside the extension (softphone/ip > phone) does not hangup, is this normal? shouldn't asterisk hangup the > extension as well when it receives the hangup properly from ISDN? or > maybe it's because asterisk does not detect the proper hangup? I don't understand your inside versus outside description. But the main point: * you have ISDN on a 3825, and you have voip. * somewhere the SIP hangup doesn't initiate an ISDN hangup. This may be (is probably) due to you having your ISDN settings incorrect for the line setup. There are about a dozen combinations of settings for the way your ISDN line could be setup. Is this a T1 PRI? Are you taking timing from the telco? Do you have your D-channel set correctly? Do you have your DSP settings correct for your chosen voip codec? You need an IOS manual to make sure all that is set correctly. You can't download one from Cisco unless you pay for support and get a service contract. Once you get that right, you'll also see the settings in the Cisco manual for pushing caller ID correctly. I think all of your problems are on the Cisco side, and zero are on the asterisk side. Or you could scrap the 3825 and get a Digium PRI card and use DAHDI. Then it would be an asterisk question :) -- _ -- 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] Opensource Speech recognition for Asterisk
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull wrote: > Voice recognition is a pain for people with accents and poor lines and when Everybody has an accent. Some people live in a place where the people they talk to sound like themselves, so they forget that fact. Of course, this is a huge problem if you, for example, want to have an English language voice recognition system that works across the continental United States. Even for people who speak 'correct' or 'common' English for their region, these systems aren't that great in my experience. The bigger of a vocabulary you have, the worse trouble you'll have, because these systems, again, in my experience, only know synonyms or alternate regional words for the same thing if they were programmed by somebody who thought of the synonyms / alternate words / alternate legitimate pronunciations. Anybody with an imagination can think of plenty examples, for example, from the United States: * soda / pop / soft drink / beverage / drink / Coke / other trademarked names -- _ -- 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] Polycom 331 freezes connecting to FreePBX
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr wrote: > We gave the phone a static IP address and pointed it to the configuration > server on the remote end that has the CFG files for it. The phone starts > up, downloads SIP and the “new application” and otherwise seems to be > booting normally. Then it gets to the “LAN Properties” screen that shows > the phone’s IP address, MAC address and firmware version and then…nothing. > It just sits there frozen. I have a suggestion... Put back the 'old application', and determine whether the 'new application' broke your phone boot. Since you don't mention changing anything else, survey says it's probably the last thing you changed that broke things. -- _ -- 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 on AMD
2010/8/13 Lyle McKarns : > Does anyone have any feelings one way or the other about running Asterisk on > AMD vs running Asterisk on Intel? Only political feelings. I want to support AMD so there's at least some token competition for Intel. Both companies make nice 6-core processors with a lot of cache. -- _ -- 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] 4 Port FXO interface
On Fri, Aug 13, 2010 at 11:43 AM, Eric Merkel (Mail Lists) wrote: > > > I am looking to build a small PBX for an office that has 3 incoming analog > lines and less than 10 extensions. For that small of an installation you might prefer an asterisk appliance. You can review the archives, or ask for recommendations. Some people like Switchvox, and there are several other 'asterisk in an appliance' options out there. -- _ -- 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 set up Asterisk to deliver a trunk sip connection?
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal wrote: > We need to upgrade this PBX for it to work with SIP, it is at the moment > using ISDN. And those who delivered it and do the > support/reconfiguration is paid by the hour. We don't have any control > over it our self, so when it is changed it will stay that way. If you have a spare ISDN card, you may prefer to integrate it over T1/E1/PRI rather than over SIP. The bigger barrier is you not having control of the system. It's made a lot tougher by not having a way to experiment except to be dead-in-the-water. -- _ -- 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 on Vmware
On Wed, Aug 11, 2010 at 4:36 AM, Tino wrote: > Is it possible to install Asterisk on Vmware(centos) from source. Is there > any difference or disadvantage for this compared to asterisk running on > physical machine. This has come up repeatedly on the list. Basically, the less you use it, the less likely you are to run into problems. Some people have reported voice quality issues, due to unreliable timing sources. Other people report they haven't seen those problems. If voice quality issues are a deal breaker for you, don't do this. -- _ -- 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 set up Asterisk to deliver a trunk sip connection?
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal wrote: > I'm trying to set up an "old" PBX (that supports SIP) to go through our > new Asterisk server, so that our old phones can be used still for some > time. > > How can I set up Asterisk to deliver a trunk sip connection that our old > PBX can connect to? Is it just to sett up a normal sip device in > sip.conf? Or is there some other / extra magic for this to work? Pretty much. The details vary on codec / DTMF, etc. based on what you're talking to, but that's the general idea. > I can't test this with the old system before we go live with Asterisk > (and then it must work). Really? Why not? If it speaks SIP, it should be able to do multiple SIP trunks / channels, and you should be able to set up a simultaneous SIP trunk alongside your production line(s). If you tell us the PBX, somebody here has probably worked with it. If the old PBX speaks true SIP, you could ditch the old PBX, and have the SIP phones register directly with asterisk. -- _ -- 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] MeetMe VS. Conference
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun wrote: > hi, group > there are two module can used for meeting. MeetMe and > Conference(which is a plugin) > > My question is : > > which is better for large conference(maybe above 100 people in a meeting)? There's at least one more choice, which is ConfBridge(), assuming you're running 1.6.2.* I personally haven't used anything except MeetMe, and I have no experience with that large of a conference. -- _ -- 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 RAID
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed wrote: > Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with > four HD's available, using CentOS as the OS. > > What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Not really an asterisk question. Asterisk will run well regardless of what you choose. -- _ -- 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] SEPMAC.xml for Ciscp 7970 IP Phone
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim wrote: > Hi All, > > I upgraded 7970 from SCCP to SIP. But the phone isn't registering. > Have you got any working XML file for 7970 phones. Isn't registering with what? If you're registering that with CallManager, you have to change the phone config after your firmware change. If you're registering with asterisk, you have to tell the phone where to try registering. -- _ -- 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] Clustering concept
On Thu, Jul 29, 2010 at 5:04 PM, wrote: > Do you know if it is possible to interconnect 1.6 with Microsoft Office > Communications Server 2007 and use the Office > Communicator as a softclient for telephone calls and the Communicator for > Instant Messaging? I believe you can set up a mediation > > server within MOC but i don't know if this works in real life. Maybe someone > here has this setup running? What I also wonder > > is if the state in Communicator client would change to "in a call" if a call > is made from Communicator via Asterisk. AFAIK MOC itself cannot > > directly communicate with a PSTN so Asterisk could be a perfect > supplemental. No, not until Microsoft builds a compatible soft phone. Microsoft built software that only speaks SIP over TCP. Most SIP stacks work over RTP. -- _ -- 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] ignorant question about Digium cards and MeetMe
So historically I've done one of two things on systems where I've needed to use MeetMe * used a real Digium card, and I've only ever used a TE400 or a TE420 for that purpose, and I know they have the timing chip * used dahdi_dummy, which works well with light load, but I had it running on a very overloaded server and had audio quality issues. I may have had quality issues even with a real Digium card, but it was impossible to isolate the problem on an overloaded server So here's my actual question... When I go through the Digium phone cards on the website, * TE420 mentions 'synchronization' between channels * TE220 mentions 'synchronization' between channels * TE121P does not mention 'synchronization' Does this mean that TE220 is the 'minimum' Digium phone card that still provides the timing / synchronization circuitry that MeetMe uses for mixing? -- _ -- 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 can conect Cisco Unified Communications Manager with Asterisk
On Thu, Jul 29, 2010 at 7:22 AM, Nguyen Quang Tri wrote: > Hello, > > i have Cisco Unified Communications Manager with 10 ip phone,i dont buy > license IVR of Cisco Unified Communications Manager. Can i use feature IVR > on Asterisk connect with Cisco Unified Communications Manager. Yes, and no. It depends on exactly how you plan to do this. You don't mention the version of your Cisco Call Manager, and you don't mention how you plan to integrate your IP phones. I'll assume Call Manager 7.x, and Cisco IP phones that support a SIP firmware. Your best way to do this: * have your Cisco IP phones running a SIP load * use SIP integration with asterisk * depending on how you want to bounce calls between Cisco and Asterisk, sometimes you should do a Transfer() rather than a Dial() to get a call from Asterisk to Cisco. It's somewhat confusing, and I have good ideas on why this is. Just trust me for now. * don't expect miracles with regards to integration of fine detailed information between Cisco and asterisk, with regard to information picked off from the IVR If your real goal is to have the same level of IVR detail from asterisk you would get from an IVR built natively within Cisco, well, sorry but that's either not possible, or a more difficult integration job than I'm familiar with. Cisco does support something called GED-125, which theoretically is an API for sending Cisco details picked off in an IVR. I'm going to play with it later in 2010. If somebody else is familiar with trying to do that thing, please speak up. -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce wrote: > :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's > not good for Queues at all. It's using the reporting tool from Areski and One of the several things you asked for was GUI for cdr database logs. FreePBX is good for putting a gui on top of doing database cdr log searching. This of course, assumes your colleagues know enough about using a database, to be able to use a pull-down web form to do filtering, and also know enough to not keep clicking when they ask for such a broad query that it takes several seconds for the database to return the results. -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce wrote: > I seem to not be able to find any good open source Asterisk Queue Analyzer > and Asterisk Log Analyzer on the web. google 'freepbx' It does some of what you want. For the rest of what you want, strongly consider paying a professional consultant. -- _ -- 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] POE Splitters
On Fri, Jul 23, 2010 at 8:46 AM, Matt wrote: > It's not necessarily this simple. There is an approximately 50-75foot cable > run through ceilings and walls (CAT5) to the location where the phones will > be. At the phone location there is no power. You always have options. You just have to decide what is more difficult: * moving the phone/devices somewhere else. Easiest solution. * having an electrician pull AC power to the location, then use DC power bricks or PoE switch * having a data cable person pull more ethernet to the location If you already have one ethernet cable that managed to make that 50-75 foot run, then clearly it can be done, and a professional could even use that cable to yank three more along the same run, and then you're all set. -- _ -- 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] POE Splitters
On Thu, Jul 22, 2010 at 2:46 PM, Matt wrote: > I've got an interesting situation where I have one cable run from the feed > area to the service area. I have three devices that I need to power at the > service area. Is anyone aware of a device that will take the POE from the > cable run and then allow me to split it to two or three devices at the > service end? The obvious answer is "don't do that". *buy DC power bricks for the phones / devices *buy a small PoE switch for the area, plugged into the single ethernet cable as a trunk *pull more cable from the original endpoint Any of those three will be more reliable and predictable when debugging than inventing your own PoE solution. I've tried to invent my own PoE solution using a soldering iron and bulk ethernet cable. Take it from me, don't go down that road. Yes, you will learn all manner of interesting things about DC voltage loss over distance, blah, blah, blah. Your time is almost undoubtedly worth more money than you'll save by pursuing the 'conventional approaches'. Just don't do it. -- _ -- 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] Audacity settings for Asterisk sound files
On Sat, Jul 17, 2010 at 6:52 PM, David Shauger wrote: > Can anyone provide the settings in Audacity to create a proper wav file > without having to do additional conversion in the cli? Has to be a way to do > this with less steps. If your goal is to 'minimize steps', you should do a batch on the command line, using Sound eXchange, aka sox. http://sox.sourceforge.net/ But nobody knows what needs to be converted, nor how, without knowing your source file format. -- _ -- 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] Where can I find a minimal set of empty configuration files (SIP only)?
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman wrote: > Hello, > > After installing and learning Asterisk I found myself with a need for a > minimal set of empty configuration files with only the "must have" stuff in > order to setup a SIP only machine, is there a place to find it? Depends on how you 'installed'. If source package, then... 'make samples' sip.conf It's full of comments. -- _ -- 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] Big time system
On Fri, Jun 25, 2010 at 11:00 AM, Cary Fitch wrote: > I see some talking about TNTs in this forum. Those are 672 lines or in some > versions double that, what is used behind them to do the processing, etc. So a channelized DS3 is roughly 28*23 channels in US if you do one D-channel per PRI (other options are possible). That gets you 644 channels. You can either buy gear that terminates a channelized DS3 natively, like a Cisco AS series device to voip-ify the PSTN channels, or you can get a device like an Adtran MX2800 which breaks out the DS3 into individual T1/PRIs, which you can then terminate with a number of different technologies, including a lot of Digium cards, or you can voipify with appliances like a Cisco 3845. So you can get a lot of asterisk boxes that have native DAHDI channels, or you can put a layer in-between that adds expense, but increases routing options. That's how the DS3 works. To bundle DS3s, you generally get fiber to the premise, and demux it at your data center using equipment approved or provided by your telco of choice. If you're talking 30k channels, that's some bigger glass, which then demuxes down to OC-whatever, which eventually demuxes to lots of DS3s, but honestly I've never worked at a scale past a handful of DS3s, so there may be a vastly superior way to do things at that scale. -- _ -- 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] Big time system
On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch wrote: > But, we have an opportunity to get into a big time telecom activity. > > It would have 2000 to 30,000 user lines per city, and we would like to have > those brought back to a central location for control and because transport > can be more economical than remote site rentals, maintenance and personnel. I would say you need to make an RFP process to first negotiate your calling rate extremely low with the major vendors of the country where you're operating. If this is US, you're talking Qwest, AT&T, Verizon, and the ilk, and you negotiate an extremely low minute rate in return for giving them a guaranteed minimum revenue. And while you're at it, you ask them how they suggest you design the architecture over their national network. -- _ -- 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] when to use e1/t1 card?
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir wrote: > This is a really rookie question: when should i use TE110P ISDN PRI Card? > > -- > Necati DEMİR When you have a single PRI / BRI line you wish to terminate into an asterisk system. -- _ -- 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] ring no answer / RONA versus HangUp
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher wrote: > On Wednesday 16 June 2010 08:21:17 David Backeberg wrote: >> I know if I do not do an Answer() that the call is not yet picked up. >> However, if I do a HangUp(), is that functionally equivalent? Can you >> Hangup() a channel you never Answer() ed? > > A Hangup just returns -1, which causes the dialplan to terminate. So yes, > you can Hangup() a call you never answered. What I was really trying to determine was whether the calling side would get the same behavior (rings, but no pickup) as if the dialplan was simulating a phone where nobody was picking it up. I ended up doing a: exten => s,1,Wait(10) exten => s,n,HangUp essentially, which was good enough for me to simulate 5 seconds of no-pickup, as perceived by the caller. Thanks much. -- _ -- 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] ring no answer / RONA versus HangUp
Hello List: I'm working on a funny scenario, where I'm bouncing calls from a Cisco call center into asterisk. Cisco call center has some logic that if a customer calls in, an agent is logged into a given extension... if Cisco sends a customer call to that extension, and there is a ring with no answer after a preset amount of time, Cisco concludes the agent is unavailable, kicks the agent off the queue, and pulls the customer call back to the front of the queue for the next available agent. This is all good. My problem is that I'm trying to figure out how best to tell asterisk 'let this ring, and don't pick it up,' that is, I want to exercise that Cisco behaviour that I've described. I know if I do not do an Answer() that the call is not yet picked up. However, if I do a HangUp(), is that functionally equivalent? Can you Hangup() a channel you never Answer() ed? I know these are pretty basic questions, but I've never thought about this problem like this before. I'm going to go ahead and play, but wanted to brainstorm this on the list. -- _ -- 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 library for C/C++
On Sun, Jun 13, 2010 at 2:59 PM, Vieri wrote: > I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6 > compatible). > I've taken a look at CAGI and QUIVR but their latest code releases date back > to 2006. > I've also seen a more recent project (wildpbx) dated 2009: > http://github.com/comradeb14ck/wildpbx/tree/master/libraries/agi/c/ > > Any suggestions/recommendations for a C AGI library? > > Thanks, > > Vieri I don't have a good answer to your real question. But my experience with the AGI libraries with Perl is that the real core of AGI is consistent. You can build on your own libraries to pass arguments correctly, etc. But the mean idea of AGI should work pretty well from 1.2 to 1.4 to 1.6. Why don't you try one of those 'old' ones, and see what problems you run into. In my experience with my Perl AGI, the problems were at the highest-level 'layer' that was just simple changes like feed it arguments with commas rather than pipes. Easy changes. -- _ -- 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 inserting records into cdr
On Thu, Jun 3, 2010 at 3:56 PM, wrote: > Hi. For several months now asterisk will mysteriously stop inserting > records into cdr database. I am using mysql and the asterisk addons > 1.6.2 to accomplish this. Sometimes there is a strange error about > column names, but often there is no error, it just stops. I just have > to restart asterisk to get things going again, so I am stumped as to > what is happening, or even how to troubleshoot. I usually run in > verbosity 4, but am not seeing anything of interest. So... I've seen this before, and I can't decide how exactly the 'problem' should be attacked. In my case, I was trying to centralize cdrs from multiple systems, down to a single cohesive cdr repository. This would give me a nice single log point I could dig through. Especially helpful in situations where you use multiple asterisks for load balancing, but then want a cohesive set of logs. Anyway, in my case, I knew for a fact that my mysql db target was sometimes flaky. Sometimes the underlying i/o layer would freak out. Or there would be a network hiccup. Here's my assessment of how cdr_mysql is behaving... * asterisk sent a db insert to the mysql database for a particular cdr * but never got that insert acknowledged * there is no hard limit on when that insert should time out, so asterisk blocks on this cdr write attempt, which will now never complete because of some problem outside the control of asterisk * then there's a traffic jam of cdrs that never even attempt because the earlier cdr did not complete, and it seems like there's a design decision that cdrs should write serially, one after the other What would I like to see as the solution? That's an excellent question. The obvious fixes are: * don't have a flaky i/o subsystem on your db box * don't have a network hiccup Barring that, it would be nice if we could set an abort timer on the cdr_mysql insert attempts. This should be user tunable. For all of my systems, much less than one second should be the return time, and I'd want an abort after, say, 10 seconds. I'd want some kind of logging trap thrown, and the error logged somewhere in /var/log/asterisk or syslog, etc. What do people think? I've known about this issue for years, but I've never seen a discussion on this. Perhaps this has come up before in Mantis. -- _ -- 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] Meetmee user introduction disabled
On Thu, May 27, 2010 at 6:17 PM, Theo Band wrote: > I used to build Asterisk from source including the zaptel-dummy module. > Last year I decided to upgrade and use a yum repository. I hoped that > this would be less hassle compared to manually chasing after the latest > release, compiling etc. And after every kernel update the modules need > to be recompiled. The yum flow does it all for me using this repository: You're going to have to file a complaint / bug / fix against whoever maintains your repository, or go back to building from source. Or get a hardware card to use for timing, as that would solve your need for dahdi_dummy :) -- _ -- 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] Delay in IVR
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart wrote: > I know nothing of Trixbox but I had a problem with my own dialplan where > there was a delay with the user selecting 0 from my IVR menu. It turned > out that because my extensions all started with 0 (they were real phone > numbers), asterisk thought that the caller might be starting to type one > of the valid extensions and so waited for the timeout (digit timeout I > think) before it went further. A similar thing can happen quite easily with FreePBX, where at least in the past, the default was that every IVR had an implicit valid selection of any extension in the system, unless you unchecked the box that made that happen. It really explained why so many calls were going to a particular phone before we found that default! -- _ -- 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] Meetmee user introduction disabled
On Thu, May 27, 2010 at 4:05 AM, Theo Band wrote: > First I noted that dahdi_dummy is no longer present in > kmod-dahdi-linux-2.3.0.1-1. Not exactly true. myhost01 asterisk # lsmod | grep dahdi dahdi_dummy 5812 0 dahdi_transcode 8968 1 wctc4xxp dahdi_voicebus 42048 2 wctdm24xxp,wcte12xp dahdi 198992 24 dahdi_dummy,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 4096 2 wctdm24xxp,dahdi myhost01 asterisk # dmesg | grep dahdi dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.3.0 > Reverting back to kmod-dahdi-linux-2.2.1-1 > solved that issue, now the module is loaded again. I suppose it would. I got dahdi_dummy with 2.3.0 by analyzing how the build process worked, and doing some tricks. I could see dahdi_dummy.c was in the package but it wasn't getting built. Here's the trick. If you pull down the combined dahdi package, extract it, cd into the extracted top-level folder cd linux (which is the dahdi proper stuff) make MODULES_EXTRA=dahdi_dummy That worked for me. Do the make install too. asktest01 linux # make MODULES_EXTRA=dahdi_dummy make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/firmware' make -C /lib/modules/2.6.28.9/build SUBDIRS=/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi DAHDI_INCLUDE=/usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/include DAHDI_MODULES_EXTRA="dahdi_dummy.o " HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/linux-2.6.28.9' CC [M] /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.o Building modules, stage 2. MODPOST 31 modules CC /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.mod.o LD [M] /usr/local/src/dahdi-linux-complete-2.3.0+2.3.0/linux/drivers/dahdi/dahdi_dummy.ko make[1]: Leaving directory `/usr/src/linux-2.6.28.9' It seems that these days you need to provide extra arguments to get dahdi_dummy, and it's getting filtered by default. -- _ -- 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] [0017330] 1.6.1 and 1.6.2 + MySQL crases on ODBC Query (via func_odbc or sip realtime)
On Mon, May 24, 2010 at 7:31 AM, Marcin J. Kowalczyk wrote: > Medium load system (~300 simultaneous calls) crases few times a day. > 1.6.1.19 but then upgraded to 1.6.2.7 but it's not solving issue. > > Any idea what can be wrong/tunned? I've three times had unexplained crashes of asterisk 1.6.2.6, and they seemed to be load related. I'm doing heavy Perl AGI. I've had other things in my todo list so I haven't circled back to investigate. I know there's a way to rebuild asterisk in a way that the coredumps are actually useful. http://www.voip-info.org/wiki/view/Asterisk+debugging if the make valgrind thing is invalid advice to build a useful debuggable version of core dumps, please somebody point to the correct directions. I have not rebuilt, again because I haven't found proper time to devote to this. I have a previous iteration of my systems where I ran with about double the number of calls with 1.6.0.5 with no crashes, and if I continue to have a tight schedule I'll back up to that version. -- _ -- 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
