Re: [asterisk-users] FastAGiin Windows Server
On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada wrote: > > My problem is that I need to execute windows app using IVR in Asterisk so we What is the windows app that you cannot replace on Linux? How about wrapping THAT program with simple inputs and outputs, and build a network interface on top of it, then bounce interface calls back and forth from linux? -- _ -- 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] [Conference] Audio/Video
On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland wrote: > I'm planning of creating a speech/video conference application. This > application will provide a system to see/listen to each personn present > in the conference. > Else, do you know any other way to do this ? http://en.wikipedia.org/wiki/CU-SeeMe it was kindof a solved problem, but that's not really around anymore. these days, ichat and google chat and Ekiga do one-on-one chat well. The problem is n-to-n chat. Take a look at openmcu, and good luck. Unfortunately, the products that work well AND are turnkey generally require money, ranging from a little to literally millions for a full-featured Cisco telepresence solution. -- _ -- 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] Problems with Fax over TDM410P
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg wrote: > On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias wrote: >> What do you mean with problems on my configuration? >> This is a FXO port on zapata: >>>> signalling=fxs_ks >>>> group=0 >>>> channel => 1 >> Not a FXS...can you explain to me what were you trying to say? > > http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType > > Yep. > If you say that's an fxo port, that's a disagreement between what you > told me and what you told the DAHDI layer. > You told DAHDI it's fxs. > Try changing the config to say fxo and tell us what happens. Of course, after re-reading what I just wrote, I think I have it backwards. My advice to flip the config and see what happens still applies. Does a regular call work fine? -- _ -- 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] Problems with Fax over TDM410P
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias wrote: > What do you mean with problems on my configuration? > This is a FXO port on zapata: >>> signalling=fxs_ks >>> group=0 >>> channel => 1 > Not a FXS...can you explain to me what were you trying to say? http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#SignallingType Yep. If you say that's an fxo port, that's a disagreement between what you told me and what you told the DAHDI layer. You told DAHDI it's fxs. Try changing the config to say fxo and tell us 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] Merge .csv files
On Tue, Apr 13, 2010 at 12:56 PM, Ricardo Coelho wrote: > Hi there, > > Does asterisk keeps the master.csv open between writes? Right now I have 2 > asterisk nodes sharing every configuration file (by using a distributed > filesystem) except the master.csv files. If asterisk does not keep master.csv > file open between writes, then I can share the master.csv file between both > nodes right?If not, then any suggestions to merge both master.csv files? Yes. download asterisk-extras compile cdr_mysql setup a shared database, point both systems at that shared database. If you're going to do anything even moderately advanced with processing your csv files, you'll be glad you went ahead and put this stuff into a database. Or you can skip the cdr_mysql, but manually dump two Master.csv files into a database to play with, if you don't mind your database not continuing to update with new info. -- _ -- 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] cause 66 - Channel not implemented
On Mon, Apr 12, 2010 at 2:23 PM, Tim Nelson wrote:
> - "Vieri" wrote:
>> Hi,
>>
>> What can I make of the following log messages? Extension 7114 tries to
>> reach 6035 but gets an "unknown channel type". What does it mean?
>> (supposedly, 6035 was not busy...)
>>
>> Apr 12 13:01:01 VERBOSE[30989] logger.c: -- Executing
>> Dial("SIP/7114-b4fe1ef0", "/6035|300|") in new stack
>> Apr 12 13:01:01 WARNING[30989] channel.c: No channel type registered
>> for ''
>> Apr 12 13:01:01 NOTICE[30989] app_dial.c: Unable to create channel of
>> type '' (cause 66 - Channel not implemented)
>> Apr 12 13:01:01 VERBOSE[30989] logger.c: == Everyone is
>> busy/congested at this time (1:0/0/1)
>>
>
> You didn't specify a channel type for extension 6035 hence the empty '' and
> subsequent call failure.
>
> Update your dialplan to reflect SIP/6035 or IAX2/6035 or CARRIERPIGEON/6035...
chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade
and check it out.
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Re: [asterisk-users] Asterisk & Timezones
On Fri, Apr 9, 2010 at 3:26 PM, Aldo Bergamini wrote: > Hi all, > > I have noticed something I can't solve regarding Asterisk (latest > 1.6.0.x). > > My server is set at the GMT+2 timezone. The clock is ok (I can get the > correct time at the terminal). But today I got a call at a time where > Asterisk should have gone 'off business hours'. > > All log times are wrong by exactly 2 hours. As if Asterisk would just > sit on GMT, ignoring the GMT+2 timezone. > > I have looked around and I do not have found any information about how > to set the log/system timezone. > > The only place I remember having a reference to timezones is the > voicemail config file; but I do not get the link to 'server time'. There's system clock, and hardware clock. Whatever you get for the localtime when you do 'date' command is what you're going to get for logs from asterisk. It seems somewhere you have your system set to run in GMT, even though you don't want it to be like that. You will need to consult documentation about properly setting your clock for your timezone. The alternative is to leave your system 'broken', and change your time checks to GMT. -- _ -- 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] res fax help
On Fri, Apr 9, 2010 at 5:40 PM, Joe Freeman wrote: > I have res_fax setup and working for the most part. However, I'm seeing > some fax machines drop the connection on me - > > Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel > 'DAHDI/1-1' did not return a frame; probably hung up. > -- Channel 0/1, span 1 got hangup, cause 102 > -- Channel 'DAHDI/1-1' FAX session '20' is complete, result: > 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: > '204x98', transfer rate: '14400', remoteSID: '' > == Spawn extension (macro-fax_rcv, s, 3) exited non-zero on > 'DAHDI/1-1' in macro 'fax_rcv' > > It appears to be dropping out of my macro fax_rcv at that point and not > executing the next step in the dialplan, which is a System call to a > script that converts the tif to a pdf and emails it to the extension owner. > > My question is how do I ensure that my script is called when the far end > hangs up before the call progresses that far in the dialplan? > > My first thought is to add something like this- > > exten => h, 1, System(callmyscript.sh,arg1,arg2,arg3,arg4,argimapirate) > > to the macro, but I'm not sure if that would do it or not. > > Anyone have any thoughts? Yes. Do the conversion in the hangup side of the context. That's the only way I've ever been able to do it. My understanding is that at the conclusion of ReceiveFax(), the line is hungup, and that is correct, normal behavior. -- _ -- 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] Problems with Fax over TDM410P
On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias wrote: >> This digium card has 3 FXO ports and 1 FXS port where we have a fax >> machine >> connected! >> >> The problem is that we can receive fax very good, but we can't make any >> outbound fax call, in fact, our asterisk get freezed in this case! >> ; TDM410P >> signalling=fxs_ks >> group=0 >> channel => 1 >> >> Signalling=fxs_ks >> group=0 >> channel => 2 >> >> signalling=fxs_ks >> group=0 >> channel => 3 >> >> signalling=fxo_ks >> group=1 >> channel => 4 >> >> What should we do in order to make it work ok? we really need to put this If you really have three FXO, and one FXS, there's part of your problem. You have your zapata configured as three FXS and one FXO. I would suspect that would be a good enough reason to crash your card or whatever. -- _ -- 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] Being attacked by an Amazon EC2 ...
On Sun, 11 Apr 2010 08:09:02 +0100 (BST), Gordon Henderson wrote: > >> Look what they did to my latency, Gordon:- >> http://f8lure.mouselike.org/archived_graphs/westek.bizorg.co.uk_day10.png > >Oddly enough my latency wasn't being affected at all - however what I was >seeing was my ADSL router being cripped with 200 packets a second in & out >- to the extent that something would go "bang" inside it and it would >drop the PPPoA session and then re-start. This was an old Draytek 2600 - I >replaced it with a new Draytek 2820 and it was them fine. I replaced my old 2600 with a BT Business hub a few months ago. The log seemed say that there were loads of corected packets. The annoying thing is that I was (trying to) work at the time and I saw the LED flashing incessantly. I checked the ther Linux box and did a "netstat" and saw nothing awry, an I thought I'd done the same on the Asterisk box. Obviously I should have looked at teh log file, because it was very obvious when I looked this morning! >It's still going on - and has been since 6am yesterday - that's now 26 >hours. Hasn't restarted here yet Fingers crossed. -- _ -- 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] Being attacked by an Amazon EC2 ...
On Sat, 10 Apr 2010 22:34:28 +0100 (BST), Gordon Henderson wrote: > >Just a "heads-up" ... my home asterisk server is being flooded by someone >from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - >they're trying to send SIP subscribes to one account - and they're >flooding the requests in - it's averaging some 600Kbits/sec of incoming >UDP data or about 200 a second )-: > >This is much worse than anything else I've seen. Same her but 184.73.17.122. Look what they did to my latency, Gordon:- http://f8lure.mouselike.org/archived_graphs/westek.bizorg.co.uk_day10.png I've had bookmarks to Fail2Ban links on my desktop for a year now. Guess I'll have to do something about it. If, hypothetically, I'd put that IP into hosts.deny - would it have stopped them? -- _ -- 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-users Digest, Vol 69, Issue 16
On Thu, Apr 8, 2010 at 10:33 PM, Alan Zheng wrote: > Hello All: > > I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO > sample configure file for them. > Is anybody know how to use them, or where is the documentation for them? If you read the code for those modules, you will learn there are NO sample configuration files because they are dialplan functions. See voipinfo for functions like: ReceiveFax() and ChanSpy() -- _ -- 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] long return times from System() calls with 1.6.2.6?
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming wrote:
> David Backeberg wrote:
>
>>> I'm doing really, really innocent things, like:
>>>
>>> exten => s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
>>
>> So I did some more testing. Same dialplan, reverted to
>> asterisk-1.6.0.13, and the contexts that do these test -e calls runs
>> lightning fast. It's like maybe there's something going on where it
>> needs to run sudo or something?
>
> There was a big change in the way the ast_safe_system() API call (used
> by the System() dialplan application) works between 1.6.0 and 1.6.2;
> it's possible you are seeing a side effect of this change. If you'd like
> to experiment, open up main/app.c (in 1.6.2), search for the
> ast_close_fds_above_n() function, and in the for() loop that runs from
> 'n+1' to 'rl.rlim_cur', change 'rl.rlim_cur' to '4096'. If that changes
> the behavior, we've found the culprit, and you can open an issue on
> issues.asterisk.org so this can be investigated.
On further review, I'm having other problems with this machine. I need
more data points before I point the finger at asterisk, as it seems
that the other 1.6.2.6 machine was fine.
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[asterisk-users] long return times from System() calls with 1.6.2.6?
I've just upgraded to 1.6.2.6 on one of my test systems. I started out
happy, with some improvements in transfers to Local() channels from a
SIP channel, and much nicer verbose fax handling.
However, something is really weird when I need to do System() calls.
It was really, really weird. This was also affecting AGI, when I
needed to read system variables from asterisk into an AGI Perl script.
I have a second test system, with asterisk-1.6.2.6, and there are not
these problems with that system.
So I suspect something whacky that really probably has nothing to do
with asterisk.
It almost feels like delay in reading loopback, or running out of
available files on the system, or something like that. I'm rebooted,
and the problem did not go away.
I'm doing really, really innocent things, like:
exten => s,1,Verbose(EXTENSION is: ${EXTEN})
exten => s,n,Set(MESSAGE_PATH=/path/to/message/)
exten => s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
exten => s,n,Verbose(System call result was ${SYSTEMSTATUS})
exten => s,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?Playback(${OVERFLOW_GENERIC}))
exten => s,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?HangUp())
exten => s,n,Goto(Next_context,s,1)
That runs lightning-fast on every system, but not on this one. There
is a huge pause, like two seconds, waiting for the System() call to
return. Dead air is not cool when setting up messaging on a phone
system.
Ideas?
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Re: [asterisk-users] long return times from System() calls with 1.6.2.6?
On Thu, Apr 8, 2010 at 4:30 PM, David Backeberg wrote:
> However, something is really weird when I need to do System() calls.
> It almost feels like delay in reading loopback, or running out of
> available files on the system, or something like that. I'm rebooted,
> and the problem did not go away.
>
> I'm doing really, really innocent things, like:
>
> exten => s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts that do these test -e calls runs
lightning fast. It's like maybe there's something going on where it
needs to run sudo or something?
Took iptables down, no change.
I run asterisk as non-root.
I then tried running asterisk as root. Same problem, so that doesn't
seem to be it.
went looking around asterisk.conf, didn't see any pertinent settings,
but then again, it's been a long time since I've built 'make
examples', so maybe there's a new setting in there that controls calls
out to the system?
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Re: [asterisk-users] How to log into separate file
On Wed, Apr 7, 2010 at 10:12 PM, Pham Quy wrote: > Hi all, > > I want to have a separate file to log what i need for my dialplan > without all output from Asterisk. By this way, i can easily to trace > problems caused by my dialplan. > > How can i do that? That's honestly a pretty vague question. Any number of problems could be caused by your 'dialplan'. syslog-ng It's nice. You can tune very specific statements to go to the arbitrary file of your choice. -- _ -- 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] Split E1 ISDN service for another device.
Hi All, I know this is not specifically Asterisk related but I don't knew where else to ask for help. Does anyone know how to or if it is even possible to allocate 512kbit/s to an ISDN device from a 30B+D ISDN line. The building the office is in has a E1 30 channel service (30B+D) but we can not get any 2B+D ISDN services. I have a HDX Polycom video conferencing system that requires a 512kbit/s service. I am told to allocate 8B+D service from the 30B+D to the Polycom device.Is this even possible. I have a Digium TE121 currently install in the server that the E1 ISDN line is connected to. The Polycom has 4 by RJ45 connections for the 512kbit/s service. Any help would be appreciated. Regards David. -- _ -- 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] Cache sound files for faster processing
On Tue, Apr 6, 2010 at 12:36 AM, huu giang wrote: > > Dear List, > > Are there any way of configuring of Asterisk so it'll cache sound files in > memory, and when Asterisk receive a call, instead of loading sound files from > the disk, it will load from the memory and so Asterisk can process much more > call at a time than with faster speed it is not caching. > > Thanks, Aside from the suggestions, you could try out an SSD drive, which is both expensive compared to a traditional hard drive and very fast. -- _ -- 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] Continuous bothering message -- Remote UNIX connection disconnected
You probably have a cron job running that executes 'asterisk -rx' -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: Monday, April 05, 2010 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Continuous bothering message -- Remote UNIX connection disconnected Hi Guys, i have a small issue but bothering me, after restarting asterisk (version 1.4 running on centos) i have the following message that comes repeatedly when i am connected to the CLI: -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected does any one know how to stop this or if it's a sign of a more serious issue? i would appreciate any help, thanks! -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] convert from wav or mp3 to gsm
On Tue, Mar 30, 2010 at 4:16 PM, salaheddine elharit wrote: > Hello All > > do you have ant software in order to change the format from mp3 or wav to > gsm in order to using it in asterisk file > > > thank you so much for your help and support > > Best Regards, > > salah If you use a 1.6 series asterisk, you can build mp3 channel support, right? make menuconfig on the source tree, and add it. Or is it in extras? -- _ -- 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] Foip solution
On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl wrote: > On Monday 29 March 2010 10:15:50 am jon pounder wrote: >> Mike Diehl wrote: >> > Hi all, >> > >> > I've cross-posted this to the -users and -biz groups. Hope that's OK. >> > >> > I have a customer who REALLY needs to be able to send/receive faxes >> > reliably. I could probably get hylafax configured, but I'm not sure how >> > reliable it is. >> > >> > If it is considered reliable, would someone let me know? >> > >> > Otherwise, is there a product/service they can buy that will allow them >> > to fax to/from their computers? >> > >> > TIA, >> >> hylafax is "the standard" never had a problem with it. >> >> used to have the odd issue with a faxmodem on a fxs port from a channel >> bank, now have it on a virtual iaxmodem, no problems at all. In fact we >> have a whole bank of virtual modems and they work just fine. > > From what I'm hearing, I could experiment with hylafax, or I can try Fax for > Asterisk. If I use Fax for Asterisk, I'll need a T.38 provider since I am > strictly using SIP trunks. Any recommendations there? The easiest recommendation: * call the local phone provider, get a few analog lines, install a fax machine Done. -- _ -- 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] Metasphere?
Hi All I'm involved in discussions with my carrier right now and am wondering if anyone has interconnected Asterisk to Metasphere via SIP? Thanks Dave -- _ -- 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 fax handeling
On Wed, Mar 17, 2010 at 5:40 AM, Peter den Hartog wrote: > Hello, > I was wondering if the following was possible: > When somebody sends a fax to my direct number 0101234567105 (my extension > will be 105) is it possible that Asterisk, or an addon sees this as a fax, > and e-mail the fax to me? > So everybody with a private extension will be able to receive faxes in his > e-mailbox on his direct number. > Any pointers would be highly appreciated! > Thanks, > Peter I've seen other requests for this before. Is this a common thing people ask for? I've always thought 'send a fax with a cover sheet' to a general number was good enough. As for the rest of it, you can convert a tiff from SpanDSP into a PDF using a linux utility like tiff2pdf, and it's your call how best you want to set up an automail utility. -- _ -- 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 running on a Xen Centos Server challenge!!!
On Wed, Mar 17, 2010 at 10:16 AM, Daniel Leite de Abreu wrote: > -bash-3.2# cd dahdi-linux-complete-2.2.1+2.2.1/ > -bash-3.2# make all > make -C linux all > make[1]: Entering directory > `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux' > make -C drivers/dahdi/firmware firmware-loaders > make[2]: Entering directory > `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware' > make[2]: Leaving directory > `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware' > You do not appear to have the sources for the 2.6.18-164.6.1.el5xen kernel > installed. > make[1]: *** [modules] Error 1 > make[1]: Leaving directory > `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux' > make: *** [all] Error 2 > > > This error tells me that i don't have the sources for the kernel > 2.6.18-164.6.1.el5xen , so how can i find it? http://wiki.centos.org/HowTos/I_need_the_Kernel_Source -- _ -- 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] ExtenSpy Problem
On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote: > David Backeberg wrote: >> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: >> You didn't mention version. Could be relevant. > Apologies for not adding the version, it's 1.4.17 Yeah, that's relevant. > I will try ChanSpy to see what happens and post the results but it > doesn't really do what I need whereas ExtenSpy does (the functionality > is required for a call centre to listen in on incoming calls and they > are not the only people using the asterisk server i.e. hosted VoIP for > multiple customers and using RealTime to boot). Do it on a test machine first. ChanSpy on 1.4.17 while running MixMonitor may crash asterisk. You might want to consider upgrading to latest 1.4 while you're at it to avoid the possible crash. I do realize you want ExtenSpy because that's the way you originally planned it. I will let you know that ChanSpy works if you can come up with a clever way to demonstrate which Chan to Spy on. And yes, I use it in a call center environment, while running MixMonitor, on 1.6.0. series -- _ -- 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 be used with Ciscs media gateways
> and also to do LCR and Quality based routing of International calls? I don't know what that means. LCR = "Least Cost Routing" Routing a call based on the quality or cost of a route (PSTN term vs SIP to PSTN term vs SIP to SIP) is actually quite common. -- _ -- 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] ExtenSpy Problem
Bumping a thread without adding anything useful is annoying. If you do it again, I won't be helping. Although I have gotten quite a chuckle from your posts, it's really going to hurt when you fall from that high horse. -- _ -- 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] ExtenSpy Problem
On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: > Hi > > I'm trying to get ExtenSpy to work but it wont, I'm dialling a number > from my mobile which comes into our server and answering the number on a > particular SIP extension which all works fine. I'm then dialling an > exten from my own SIP extension which executes the ExtenSpy for the > correct extension but I hear nothing. You didn't mention version. Could be relevant. Bumping a thread without adding anything useful is annoying. If you do it again, I won't be helping. Try turning off the recording before using ExtenSpy(). You tell us what happens. Try ChanSpy() instead of ExtenSpy() Type your version number and your 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 to be used with Ciscs media gateways
On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote: > I have been trying to do this since 2 days but couldn't make itneed your > help.. Well, you could certainly ask Cisco for help. You did pay Cisco money, right? > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers > I am able to place call from cisco gateway to the asterisk box and also to > some softphones extensions but >when making a call from softphone from > asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that > the To field is SIP header is coming as sip:41.205.190.15 which is not > correct, instead it should be dialed >number:41.205.190.15 Then the problem seems to be between your asterisk box and your Cisco. Perhaps if you told us what you were trying to SIP dial, we would be able to tell us what you did wrong. > Has any one of you tried using Asterisk in this scenario yes. > and also to do LCR and Quality based routing of International calls? I don't know what that means. > Please let me know if there is any documentation /example of this kind > available There is. cisco.com you pay them, then you can use their documentation. -- _ -- 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] app_confbridge production ready?
On Fri, Mar 5, 2010 at 3:06 PM, Robert McGilvray wrote: > > Does anyone use confbridge in a large installation and can provide feedback > on its stability, quality in comparison to MeetMe? I use a sangoma card in > my 1.4.2 box to provide timing and it has never been an issue. Can I expect > similar performance from the new timing API? I do not use ConfBridge() in a large installation. I use MeetMe on 1.6.0.* The timing is different for ConfBridge, as it does not require DAHDI. If you have that good of an experience with 1.4, why change 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 SIP, SIPAddHeader() and Cisco GED-125
On Mon, Mar 8, 2010 at 1:42 PM, Franklin Webb wrote: > Hello David, > > I had an application where I had to pass data between Asterisk and a Genesys > system using SIPAddHeader(). It worked pretty well, but we had two genesys > boxes, and by the time I was done I found I was losing the SIP header where I > needed it, since it only shows up on next INVITE. I ended up storing data in > the CallerID Name field with a delimeter and parsing it out. Far from an > ideal solution, but it may be something that can help you. Thank you very much for your reply. That had occurred to me as well, and it may very well be the approach I take. I agree with you that it is less than ideal, but if a solution works at all, that's good enough for my needs. Luckily, the most important value I want to pass will fit in the callerID field width. -- _ -- 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] Free 'Locked up' Channels
I would love to see any info on this as well. I see similar issues with meetme bridges having locked channels. It's easy to set a timeout but a fix (maybe I'm just doing something wrong?) would be better than a workaround. -d -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Chamberlain Sent: Wednesday, March 03, 2010 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Free 'Locked up' Channels Hi All, Asterisk 1.4.25.1 .22 .29 - pretty much every Asterisk install we have out there exhibits this. Just wondering how to free a channel that will stay eternally busy ala: carl*CLI> core show channels Channel Location State Application(Data) SIP/101-Dotnet-09bb2 *...@from-inside-dotne Down(None) 1 active channel 0 active calls This channel is not active. But Asterisk will never free it. Unfortunately it affects SIP subscriptions so people think this extension is always busy. Restart when convenient is no use because Asterisk will always think this channel is in use. I can force a restart but I would prefer if there was a way to free this channel from the CLI. TIA. Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi not available in Asterisk
Hi All, I must be doing something really stupid as I can't get DAHDi working in Asterisk. It is loaded and working in Linux fine. *CLI> module load chan_dahdi Unable to load module chan_dahdi Command 'module load chan_dahdi' failed. [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:393 load_dynamic_module: Error loading module 'chan_dahdi': libpri.so.1.4: cannot open shared object file: No such file or directory [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:770 load_resource: Module 'chan_dahdi' could not be loaded. >From the Log file. [2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module 'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such file or directory [2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module 'chan_dahdi.so' could not be loaded. I am using on CentOS 5.4 64 bit. Asterisk1.6.0.25 Asterisk-addons 1.6.0.4 Libpri 1.4.10.2 I have install libpri first and then asterisk. Regards David. -- _ -- 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] 30 mins GSM file
Sorry if you guys find this silly, for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm >>* /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ; done* I need to enter above lines in my root prompt ? for i in `seq 1 180`; do cat /var/lib/asterisk/sounds/en/silence/10.gsm * /var/lib/asterisk/sounds/30-minutes-of-silence.gsm ;* * * * * * * On Fri, Mar 5, 2010 at 4:36 AM, David @ULC wrote: > > I believe we GSM of 8 bit for Asterisk ? > > > On Fri, Mar 5, 2010 at 4:35 AM, David @ULC wrote: > >> Record a muted channel for 30 minutes like this: >> >> exten => s,1,Answer(1) >> >> exten => s,n,Progress() >> >> exten => s,n,record(silence_long.gsm|1800|s) >> >> exten => s,n,hangup >> >> >> >> >> Above option looks easy. >> >> What I have to dial from soft phone to get this ? >> >> >> >> On Fri, Mar 5, 2010 at 4:21 AM, David @ULC wrote: >> >>> >>> I need to create 30 mins of GSM file for Asterisk . >>> >>> Silent / Blank file. >>> >>> Whats the best way to create 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
[asterisk-users] Hardware requirements question.
I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little M&M Technology, Inc. da...@mandm-tech.com 704.882.9432 x3 704.882.0405 FAX -- _ -- 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] 30 mins GSM file
I believe we GSM of 8 bit for Asterisk ? On Fri, Mar 5, 2010 at 4:35 AM, David @ULC wrote: > Record a muted channel for 30 minutes like this: > > exten => s,1,Answer(1) > > exten => s,n,Progress() > > exten => s,n,record(silence_long.gsm|1800|s) > > exten => s,n,hangup > > > > > Above option looks easy. > > What I have to dial from soft phone to get this ? > > > > On Fri, Mar 5, 2010 at 4:21 AM, David @ULC wrote: > >> >> I need to create 30 mins of GSM file for Asterisk . >> >> Silent / Blank file. >> >> Whats the best way to create 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] 30 mins GSM file
Record a muted channel for 30 minutes like this: exten => s,1,Answer(1) exten => s,n,Progress() exten => s,n,record(silence_long.gsm|1800|s) exten => s,n,hangup Above option looks easy. What I have to dial from soft phone to get this ? On Fri, Mar 5, 2010 at 4:21 AM, David @ULC wrote: > > I need to create 30 mins of GSM file for Asterisk . > > Silent / Blank file. > > Whats the best way to create 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
[asterisk-users] 30 mins GSM file
I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create 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
[asterisk-users] asterisk SIP, SIPAddHeader() and Cisco GED-125
Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of what these packets need to look like. But wouldn't it be nice if instead, you could use SIPAddHeader() with X tags and have Cisco pick off the out-of-band values from SIP packets? Wouldn't it be even nicer if there was a middleware that spoke GED-125 out of one side, and spoke SIP X headers on the other side? I will soon be able to tell you about the bowels of this interaction, but before I go down this road, does anybody want to speak up with lessons learned from doing this themselves? I'm assuming I'm going to end up creating a library in Perl to help me do this (that is, the out-of-band conversation with the CVP). -- _ -- 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 and usernum
On Mon, Mar 1, 2010 at 11:01 AM, Emrah wrote: > Hi! > > Thanks a lot for your answer. > The problem with the command you mentioned is... When do I call it? If two > people happen to enter the conf at the sametime, > I have a feeling there may be some little confusion there... > Do you think I could use the agi-background option with meetme? > I am using 1.6. You'll need to figure out the channel the caller was originally on before you dump them into meetme, then grep for that channel on the output of meetme list to figure out their number in the meetme room. I personally would fire up an agi, pick off the channel, put them in the room, then grep on the meetme list, then set / store the variable. -- _ -- 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 and usernum
On Mon, Mar 1, 2010 at 6:42 AM, Emrah wrote: > I am trying to get the usernum of a user when dialing in to a MeetMe > conference. Is there somehow a possibility to save the usernum of a > MeetMe participant into a variable? Everything should be done through > the DialPlan, no manager and no *cli. You don't say what version you're running. I second Steve's claim. Even with 1.6, I can't think of how to do what you want without resorting to AGI. Which is technically in the dialplan, but you're going to have to do extra work elsewhere. If you're using 1.6, you will enjoy knowing about 'meetme list concise', which you can then process with awk. If you absolutely don't want to do AGI, you could always modify meetme.c, recompile, and share your work with others. I think you'll find that harder than writing an AGI. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco DTMF
On Mon, Mar 1, 2010 at 9:40 AM, David Backeberg wrote: > On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs > wrote: >> Hi, >> >> I have encountered a DTMF issue. My scenario: >> >> Access carrier-sip> >> Asterisk-1.4.25.1-sip>CiscoGW-ISDN->TDM Switch >> >> the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk >> forwards it with SIP INFO method to Cisco gateway, but on TDM switch every >> digit is duplicated. Is it possible that the carrier sends inband along with >> rfc2833? > Can you take asterisk out of the loop, terminate sip carrier straight > into Cisco for testing? You could also make a really simple dialplan object to do some DTMF directly with a channel on the asterisk, to see if things work properly going just that far. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco DTMF
On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs wrote: > Hi, > > I have encountered a DTMF issue. My scenario: > > Access carrier-sip> > Asterisk-1.4.25.1-sip>CiscoGW-ISDN->TDM Switch > > the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk > forwards it with SIP INFO method to Cisco gateway, but on TDM switch every > digit is duplicated. Is it possible that the carrier sends inband along with > rfc2833? Possible? Sure. Also possible that Cisco is passing along the in-band, as well as converting the out-of-band to in-band, ergo two for one. You can also tune the DTMF on the Cisco to ignore or set parameters on DTMF. Refer to the IOS guide for the appropriate arguments. Even worse, it's possible that you have a lot of echo, and the DTMF is echo-y enough that it gets interpreted as two-for-one. Can you take asterisk out of the loop, terminate sip carrier straight into Cisco for testing? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk RPM's
On Fri, Feb 26, 2010 at 11:24 AM, Jay Vocaire wrote: > I am new to Asterisk and have searched all over for an answer to this, > so please don't skewer me too bad if this is a stupid question. I am > currently running 1.6.0.21 on a few test boxes (one i386, one x64), and > have noticed that there haven't been any RPM updates since .21, even > though .25 just hit. > > What I am wondering about (and please don't assume this is a complaint, > I simply don't know the reasoning) is the lag between the release of the > version and the RPM availability. Is there something that needs to be > done other than compiling the code? If so, what is it? I'm assuming that somebody, somewhere built a SPEC file, which is (I thought) required to build an rpm. If you can track that down, you should be able to use that spec file and change out the source it's pointing to, to build your own rpms at will. I just did a find | grep -i spec on an asterisk source tree, so I'm going out on a limb and saying it's not distributed in the normal source package. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Morse Code
Does anybody use the Morsecode app for anything interesting? I'm strangely fascinated by this core piece of Asterisk functionality. Duh! How are we going to spread the word about how to take those alien bastards down if we don't keep morse code around!?!??! http://www.imdb.com/title/tt0116629/ [quote] 02:17:03 We know how to take 'em out, General. Spread the word. 02:17:10 Get on the wire to every squadron around the world. 02:17:14 Tell them how to bring those sons of bitches down. [/quote] :) -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD: HANGUP
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC wrote:
> *Code:*
>
> == Manager 'sendcron' logged off from 127.0.0.1
> -- Executing Playback("Local/91441425477...@default-b9f2,1",
> "sip-silence") in new stack
> -- Playing 'sip-silence' (language 'en')
> -- Executing AGI("Local/91441425477...@default-b9f2,1", "agi://
> 127.0.0.1:4577/call_log") in new stack
> -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
> -- Executing AMD("Local/91441425477...@default-b9f2,1",
> "2000|2000|1000|5000|120|50|4|256") in new stack
> -- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt:
> 64)
> -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
> [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
> [50] maximumNumberOfWords [4] silenceThreshold [256]
> == Spawn extension (default, 91441425477375, 2) exited non-zero on
> 'Local/91441425477...@default-1e22,2'
> -- Executing DeadAGI("Local/91441425477...@default-1e22,2", "agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15")
> in new stack
> -- AGI Script agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
> returning 0
> -- AMD: HANGUP
> -- Executing DeadAGI("Local/91441425477...@default-1e22,1", "agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---")
> in new stack
> -- AGI Script agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
> returning 0
> == Spawn extension (default, 91441425477388, 2) exited non-zero on
> 'Local/91441425477...@default-86e4,2'
> -- Executing DeadAGI("Local/91441425477...@default-86e4,2", "agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15")
> in new stack
> -- AGI Script agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
> returning 0
> -- AMD: HANGUP
> -- Executing DeadAGI("Local/91441425477...@default-86e4,1", "agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---")
> in new stack
> -- AGI Script agi://
> 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
> returning 0
> vici*CLI>
>
>
>
> My agent are NOT getting calls.
>
> -- AMD: HANGUP ??
>
> Is that an Issue ?
>
> How to solve it ?
>
>
> I have below entry for 8369 :
>
> *Code:*
> ; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
> exten => 8369,1,Playback(sip-silence)
> exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
> exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
> exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
> exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
> exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
> exten => 8369,7,Hangup
>
>
> Amd.conf has :
>
> *Code:*
>
> ; initial_silence: Maximum silence duration before the greeting. If
> exceeded then MACHINE.
> ; greeting: Maximum length of a greeting. If exceeded then MACHINE.
> ; after_greeting_silence: Silence after detecting a greeting. If exceeded
> then HUMAN
> ; total_analysis_time: Maximum time allowed for the algorithm to decide on
> a HUMAN or PERSON
> ; min_word_length: Minimum duration of Voice to considered as a word
> ; between_words_silence: Minimum duration of silence after a word to
> considere the audio what follows as a new word
> ; maximum_number_of_words: Maximum number of words in the greeting. If
> exceeded then MACHINE
>
>
> [AnsweringMachineDetector]
> initial_silence= 3500
> greeting = 1500
> after_greeting_silence = 300
> total_analysis_time= 5000
> min_word_length= 120
> between_words_silence = 50
> maximum_number_of_words= 5
> silence_threshold = 256
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[asterisk-users] AMD: HANGUP
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/91441425477...@default-b9f2,1",
"sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI("Local/91441425477...@default-b9f2,1", "agi://
127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD("Local/91441425477...@default-b9f2,1",
"2000|2000|1000|5000|120|50|4|256") in new stack
-- AMD: Local/91441425477...@default-b9f2,1 00 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence
[1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
[50] maximumNumberOfWords [4] silenceThreshold [256]
== Spawn extension (default, 91441425477375, 2) exited non-zero on
'Local/91441425477...@default-1e22,2'
-- Executing DeadAGI("Local/91441425477...@default-1e22,2", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15")
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-35-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI("Local/91441425477...@default-1e22,1", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---") in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
== Spawn extension (default, 91441425477388, 2) exited non-zero on
'Local/91441425477...@default-86e4,2'
-- Executing DeadAGI("Local/91441425477...@default-86e4,2", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15")
in new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-16-ANSWER-41-15completed,
returning 0
-- AMD: HANGUP
-- Executing DeadAGI("Local/91441425477...@default-86e4,1", "agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---") in
new stack
-- AGI Script agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-0---completed,
returning 0
vici*CLI>
My agent are NOT getting calls.
-- AMD: HANGUP ??
Is that an Issue ?
How to solve it ?
I have below entry for 8369 :
*Code:*
; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-LB)
exten => 8369,7,Hangup
Amd.conf has :
*Code:*
; initial_silence: Maximum silence duration before the greeting. If exceeded
then MACHINE.
; greeting: Maximum length of a greeting. If exceeded then MACHINE.
; after_greeting_silence: Silence after detecting a greeting. If exceeded
then HUMAN
; total_analysis_time: Maximum time allowed for the algorithm to decide on a
HUMAN or PERSON
; min_word_length: Minimum duration of Voice to considered as a word
; between_words_silence: Minimum duration of silence after a word to
considere the audio what follows as a new word
; maximum_number_of_words: Maximum number of words in the greeting. If
exceeded then MACHINE
[AnsweringMachineDetector]
initial_silence= 3500
greeting = 1500
after_greeting_silence = 300
total_analysis_time= 5000
min_word_length= 120
between_words_silence = 50
maximum_number_of_words= 5
silence_threshold = 256
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Re: [asterisk-users] Problems in Asterisk Real Time (Urgent help )
Have you check if MySql is already running? Have you check HD space? regards. 2010/2/24 ahmed magdy > Hello, > > Asterisk Real time database worked on astersik 1.6.2.0 but now i am working > on Asterisk to latest version which is 1.6.2.2 ,there is a a warning > [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime > mapping for 'sippeers' found to engine 'mysql', but the engine is not > available > [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register: > Registration from '"555">' > failed for '192.168.50.105' - No matching peer found > > is there a problem in version compatability? > > if anyone knows anything ,help me please. > > -- > > Ahmed Magdy Mahmoud > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Juan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] subject: 1.4 vs 1.6
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro wrote: > Hi Guys > > > We are using asterisk 1.4 on all of our platforms for a while now. > Some of our partners recommended to use asterisk 1.6 in order to improve > overall stability and performance. > > Can someone please let me know if you have a such experience? > Also, do you have any other negative or positive comments on 1.6 If it isn't broke, don't 'fix' it. There are benefits to 1.6, like dramatically enhanced SIP support, much faster dialplan processing, easier faxing, changes to dialplan syntax, and lots of other features. I would say the improvement of going to 1.6 is only if you are trying to expect more from the same gear, or want the new features. If you're not actually having problems, don't change anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open source or low-budget recommendation for call-center software
I think Vicidial, works great. Regards. 2010/2/22 Apa Minerala > Hello, > > We used to recommend a commercial software but client is a small callcenter > who cannot afford something big. > > Would you recommend something open-source which could work for a 40-seater? > > > Thank you, > > Tudor > > www.sunabasarabia.com > Moldova 11c/min > Romania 2c/min > $1 de test de la bun inceput > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Juan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem w/ MoH
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl wrote: > Hi all, > > I'm trying to get moh working on * version 1.4.4. I've setup a test I don't know the answer, but are you really using 1.4.4? If so, consider taking some time to review the security and feature improvements over the last several years and seriously consider updating. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman wrote: > David Backeberg wrote: >> Timers are built on the premise that they have access to either a real >> timing device, or unobstructed access to a processor which clicks >> through a proc cycle at a pre-determined rate. Once you break those >> rules, don't be surprised when the timers stop working, and 'bad >> things' happen. > > Forgive the possibly stupid question, but do these problems you describe > apply equally to the dom0 as to any domU's in a xen system? I used to > think not, but now I'm starting to realize that I'm probably mistaken... http://wiki.xensource.com/xenwiki/Scheduling It sounds like there are multiple ways to do scheduling in a Xen situation. The best way to avoid overloading the system is to deliberately underutilize the system, but then what's the point of virtualization? The supposed benefits of virtualization are power savings, and better utilization of existing resources. If you're using it for other reasons like a development environment, you'll probably be fine. To be clear, you may get away with virtualization and never run into any problems. But you have to know who to blame when you DO run into problems. Having problems of the sort uniquely caused by starving virtual kernels for resources is not going to be the fault of asterisk, but rather a failure to anticipate the downside of trying to use virtualization with asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual machine timing (KVM)
On Sun, Feb 21, 2010 at 10:04 PM, Sean Brady wrote: > I do get choppy audio when playing recordings occasionally. I haven’t had > time to figure that one out, but I haven’t put it into production yet. You just said you're getting unexplained choppiness. You also just said you're not in production. > I have been told repeatedly that Asterisk shouldn’t be virtualized, and that > timing was an issue, however I have never been given a reason that I > consider acceptable to preclude me from doing so. How about the fact you're getting unexplained choppiness before you're even in production? > surrounding Asterisk virtualized. Perhaps I am just stubborn, but I am > determined to run Asterisk virtualized in production with conferencing (be > it meetme or confbridge) until it’s been proven without doubt that it just > doesn’t work. What exactly would constitute 'proof without a doubt' that would satisfy you? If your virtualized webserver has to fight it out with other virts, and your webserver takes an extra second to process a web page, not such a big deal. If that's your audio conference that just had to spin for a second, you just lost words out of a sentence. If it happens during authentication, you dropped digits and the auth fails. If it happens during call setup, the call might not go through. If it happens during hangup, the hangup might get missed. UDP does NOT retransmit. Get it? Now do you understand why it's a bad idea? Timers are built on the premise that they have access to either a real timing device, or unobstructed access to a processor which clicks through a proc cycle at a pre-determined rate. Once you break those rules, don't be surprised when the timers stop working, and 'bad things' happen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 4 PCIe cards in one asterisk server
On Mon, Feb 22, 2010 at 8:20 AM, Arjan Kroon | Mobillion wrote: > Hi, > > > > Does anybody have any experience with asterisk where are four PCIe cards are > used in one server (TE420). > > So you can have max 4 * 4 * 30 channels = 480 channels used. I would recommend calling Digium and asking them. They may have particular models that are known to work in that configuration. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IPKall NOT coming on Asterisk
Its crazy I made it working . Today I had to reinstall all due to soem reason. Now, when I am trying, its NOT coming. Same CPU, Same Lan, Same Windows which acts as Internet Gateway. CALL "Doesnt" hit my Asterisk. http://i50.tinypic.com/1z3axrc.jpg http://i45.tinypic.com/23mr5uq.jpg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd wrote: > How much control do the ssh processes have over the call, if any? It occurred to me that I might be answering this backwards. So from the perspective of server A, trying to talk to a remote system B running asterisk, server A can invoke: asterisk -rx "do something on the asterisk cli" and it will be done to asterisk on that system. So for example, I have built a nifty web gui that displays current call status in the system, along with a bunch of buttons. Among those buttons, is one that will hangup a call, on an appropriate channel, as corresponds to the database state I've been maintaining. And this does happen to have been done in PHP. And to do this hangup, I actually do NOT run ssh with keys, but rather I use asterisk manager. And send I use a nice PHP Asterisk manager library that somebody else wrote to manage the connection, then I send the Hangup() command on the appropriate channel, and the PHP Asterisk manager takes care of the dirty work of closing and cleaning up the connection. I chose to use PHP and asterisk manager, but I could have done the same thing with ssh keys and asterisk -rx '' approach. Hopefully that helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd wrote:
> Hello David,
>
> Thanks so much for your message!
>
> Please check my comments inline below...
> David Backeberg wrote:
>> On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd wrote:
>>
>>> Hello there,
>>>
>>> I'm trying to figure out how to run a PHP script on a remote machine and
>>> still have access to the audio stream associated with the call.
>>>
>>> Ideally, I'd love to play/record audio files directly from/to the remote
>>> server without having to copy them back and forth to the Asterisk
>>> server. What is the best way to do this?
>>>
>> 1) recordings, with a side order of distributing those to another machine
>> 2) remote shell scripting
>>
> What would be the "asterisk way" of recording part of the call from a
> remote server? I'm not sure I can do that (the remote connection) with
> EAGI, can I?
The 'asterisk way' of recording part of a call needs to be done on the
asterisk system where the call is taking place. Or, if the call is
actually between two asterisk systems, the call can be recorded
directly on one or the other or both asterisk systems. The asterisk
dialplan feature is called Monitor() or MixMonitor(), and you can
refer to the documentation for the differences.
The AGI and remote connection comes in when the recording (call)
completes, and in the h (hangup) context for this dialplan context,
you would do the remote file copy so your call would now be copied off
somewhere else.
> Do you know of any examples that use ssh from inside Asterisk calls?
Sure, here's an example from one of my dialplans.
exten => s,1,Answer
exten => s,n,Set(CDR(userfield)=faxsample)
exten => s,n,Set(LOCALSTATIONID=FaxSample)
exten => s,n,Set(LOCALPATH=/var/spool/fax/recvq/${LOCALSTATIONID}/)
exten => s,n,Set(MYDATE=${STRFTIME(${EPOCH},,%C%y-%m-%d-%H%M)})
exten => s,n,Set(MYFILENAME=${LOCALSTATIONID}-${MYDATE}-${CDR(uniqueid)})
exten => s,n,Set(MYFULLPATH=${LOCALPATH}${MYFILENAME})
exten => s,n,Set(KEYFULLPATH=/var/spool/fax/ssh_key_for_remote_copy)
exten => s,n,Set(SCPUSER=filecopyuser)
exten => s,n,Set(FILESERVER=fileserver.domain.com)
exten => s,n,Set(REMOTEPATH=/path/to/where/it/should/go/${LOCALSTATIONID})
exten => s,n,Set(RECORDING=${LOCALPATH}recording/${MYFILENAME}.gsm)
exten => s,n,MixMonitor(${RECORDING})
exten => s,n,Playback(silence/1)
exten => s,n,ReceiveFax(${MYFULLPATH}.tif)
; log what happened with the fax transmission
exten => h,1,System(/bin/echo ${MYFULLPATH},${CALLERID(num)},${CALLERID(name)},$
{FAXSTATUS},${FAXERROR},${FAXMODE},${FAXPAGES},${REMOTESTATIONID} >>
${LOCALPATH}fax.log)
;
; if fax is salvageable, a tif will exist.
exten => h,n,System(test -e ${MYFULLPATH}.tif)
;exten => h,n,Verbose(System call result was ${SYSTEMSTATUS})
;
; try to turn any file that exists into a pdf
exten => h,n,ExecIf($[${SYSTEMSTATUS} =
SUCCESS]?System(/usr/bin/tiff2pdf -p"letter" ${MYFULLPATH}.tif -o
${MYFULLPATH}.pdf))
;
; check if pdf exists. If the fax was too incomplete to process, no
file will exist.
; If yes, send it off to H drive
exten => h,n,System(test -e ${MYFULLPATH}.pdf)
;exten => h,n,Verbose(System call result was ${SYSTEMSTATUS})
exten => h,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?System(/usr/bin/scp
-i ${KEYFULLPATH} ${MYFULLPATH}.pdf
${scpus...@${fileserver}:${REMOTEPATH}))
; if tiff file exists from good fax, name it one thing
exten => h,n,ExecIf($[${FAXSTATUS} = SUCCESS]?System(/bin/mv
${MYFULLPATH}.tif /var/spool/fax/recvq/processed))
; if fax was bad, check if we still have tif. If so, move it out
exten => h,n,System(test -e ${MYFULLPATH}.tif)
;exten => h,n,Verbose(System call result was ${SYSTEMSTATUS})
exten => h,n,ExecIf($[${SYSTEMSTATUS} = SUCCESS]?System(/bin/mv
${MYFULLPATH}.tif
/var/spool/fax/recvq/processed/${MYFILENAME}-FAILED.tif))
exten => h,n,Hangup
> How much control do the ssh processes have over the call, if any?
As you can see in that particular example, I was using scp to do a
remote file copy of the received fax. I also setup MixMonitor()
against the channel with a filename that would match the cdr of the
call in case I ever needed to go back and troubleshoot a particular
fax.
> Is that comparable to Fast_AGI? Or EAGI?
Ummm, kindof. My example shows doing everything directly in asterisk
dialplan. AGI let's you use the language you prefer to do arbitrary
things with calls, using the AGI library for that language. Some
people prefer AGI, some people prefer dialplan. They both have their
strengths, and drawbacks.
Strength of dialplan is it's extremely debuggable. Strength of AGI is
that you get to leverage the syntax and libraries in a language you
already use, but when you want to debug, you have (in my opinion) less
introsp
Re: [asterisk-users] Virtual machine timing (KVM)
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti wrote: > To get MeetMe working properly, I know some sort of timing device > provided by the zaptel package is required (even if it means the > zt_dummy). But, on a virtual machine I know that the Linux timing won't > work as expected. Is it possible to then dedicate a physical device > like a USB port or something to the virtual machine to use for the > timing interrupts? You could always use ConfBridge(), starting in 1.6.2.*, which does not require DAHDI/Zaptel, and therefore doesn't require a timer. Let me be the first to tell you that using a virt for a conferencing solution, especially if you want people to actually use it, sounds like a 'Bad Idea'. You could oversubscribe the resources so you don't starve the virt, but we already have a name or that. It's called not using a virt in the first place. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static IP
Feb 17 20:42:17 NOTICE[2746]: chan_sip.c:5529 sip_reg_timeout:--
Registration for '11012012...@proxy.ideasip.com' timed out, trying again
(Attempt #119)
-- parse_srv: SRV mapped to host proxy.ideasip.com, port 5060
-- Got SIP response 479 "Please don't use private IP addresses" back
from 208.97.25.11
I cant use Ideasip ???
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC wrote:
>
> So, this will change :
>
> register => 11012012600:passw...@proxy.ideasip.com/11012012600
>
> [ideasip]
> type=friend
> secret=password
> username=11012012600
> host=proxy.ideasip.com
> insecure=very
> fromdomain=proxy.ideasip.com
>
> exten => _1101XXX,1,SetCallerID(Your Name <11012012600>)
> exten =>
> _1101XXX,2,Dial(SIP/${ext...@proxy.ideasip.com
> )
> exten => _1101XXX,3,Hangup
>
>
> Feb 17 20:42:17 NOTICE[2746]: chan_sip.c:5529 sip_reg_timeout:--
> Registration for '11012012...@proxy.ideasip.com' timed out, trying again
> (Attempt #119)
> -- parse_srv: SRV mapped to host proxy.ideasip.com, port 5060
> -- Got SIP response 479 "Please don't use private IP addresses" back
> from 208.97.25.11
>
>
>
>
>
> On Thu, Feb 18, 2010 at 5:34 AM, David @ULC wrote:
>
>>
>> hmmm Ok..
>>
>> Is this a Asterisk Question ?
>>
>> I have a setting as :
>>
>> Global Settings :
>> ---
>> SipIpkall = SIP/fwd
>>
>> Dialplan Entry :
>>
>> exten => 11012012600,1,Ringing call ringing
>> exten => 11012012600,2,Wait(1) Wait 1 second for CID delivery from
>> PRI
>> exten => 11012012600,3,Answer Answer the line
>> exten =>
>> 11012012600,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-11012012600-Closer-park--999-1-TESTCAMP)
>> exten => 11012012600,5,Hangup
>>
>>
>> Registration String: :
>> ---
>> register
>> =>11012012600:passw...@66.54.140.46<11012012600%3apassw...@66.54.140.46>
>>
>>
>> Sip Entry :
>> --
>> [fwd]
>> type=friend
>> secret=password
>> username=11012012600
>> host=66.54.140.46
>>
>>
>>
>> I get in CLI ::
>>
>> Feb 17 19:03:17 NOTICE[2554]: chan_sip.c:5529 sip_reg_timeout:--
>> Registration for '11012012...@66.54.140.46' timed out, trying again
>> (Attempt #18)
>>
>> When I try to Ping from my CentOS , I can ping 66.54.140.46.
>>
>>
>> On Thu, Feb 18, 2010 at 3:11 AM, David @ULC wrote:
>>
>>>
>>> Looks like IdeaSip need STATIC ip else it doesnt work.
>>>
>>> .
>>>
>>>
>>> On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote:
>>>
>>>> Ok
>>>>
>>>> I can use
>>>>
>>>> Dyndns.org
>>>>
>>>> I registered myself.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> easy.selfip.com
>>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>
>>>> successfully activated.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d>
>>>>
>>>>
>>>>
>>>> Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a>
>>>>
>>>>
>>>>
>>>> Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a>
>>>>
>>>>
>>>>
>>>> Last Updated
>>>> <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com
>>>>
>>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline
>>>>
>>>>
>>>>
>>>> 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM
>>>>
>>>>
>>>> What next ???
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote:
>>>>
>>>>> I dont have a Static IP.
>>>>>
>>>>> How can I ask IPKall to send call to my Asterisk ?
>>>>>
>>>>>
>>>>
>>>
>>
>
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Re: [asterisk-users] Static IP
http://i50.tinypic.com/120rwya.jpg
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC wrote:
>
> So, this will change :
>
> register => 11012012600:passw...@proxy.ideasip.com/11012012600
>
> [ideasip]
> type=friend
> secret=password
> username=11012012600
> host=proxy.ideasip.com
> insecure=very
> fromdomain=proxy.ideasip.com
>
> exten => _1101XXX,1,SetCallerID(Your Name <11012012600>)
> exten =>
> _1101XXX,2,Dial(SIP/${ext...@proxy.ideasip.com
> )
> exten => _1101XXX,3,Hangup
>
>
> Feb 17 20:42:17 NOTICE[2746]: chan_sip.c:5529 sip_reg_timeout:--
> Registration for '11012012...@proxy.ideasip.com' timed out, trying again
> (Attempt #119)
> -- parse_srv: SRV mapped to host proxy.ideasip.com, port 5060
> -- Got SIP response 479 "Please don't use private IP addresses" back
> from 208.97.25.11
>
>
>
>
>
> On Thu, Feb 18, 2010 at 5:34 AM, David @ULC wrote:
>
>>
>> hmmm Ok..
>>
>> Is this a Asterisk Question ?
>>
>> I have a setting as :
>>
>> Global Settings :
>> ---
>> SipIpkall = SIP/fwd
>>
>> Dialplan Entry :
>>
>> exten => 11012012600,1,Ringing call ringing
>> exten => 11012012600,2,Wait(1) Wait 1 second for CID delivery from
>> PRI
>> exten => 11012012600,3,Answer Answer the line
>> exten =>
>> 11012012600,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-11012012600-Closer-park--999-1-TESTCAMP)
>> exten => 11012012600,5,Hangup
>>
>>
>> Registration String: :
>> ---
>> register
>> =>11012012600:passw...@66.54.140.46<11012012600%3apassw...@66.54.140.46>
>>
>>
>> Sip Entry :
>> --
>> [fwd]
>> type=friend
>> secret=password
>> username=11012012600
>> host=66.54.140.46
>>
>>
>>
>> I get in CLI ::
>>
>> Feb 17 19:03:17 NOTICE[2554]: chan_sip.c:5529 sip_reg_timeout:--
>> Registration for '11012012...@66.54.140.46' timed out, trying again
>> (Attempt #18)
>>
>> When I try to Ping from my CentOS , I can ping 66.54.140.46.
>>
>>
>> On Thu, Feb 18, 2010 at 3:11 AM, David @ULC wrote:
>>
>>>
>>> Looks like IdeaSip need STATIC ip else it doesnt work.
>>>
>>> .
>>>
>>>
>>> On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote:
>>>
>>>> Ok
>>>>
>>>> I can use
>>>>
>>>> Dyndns.org
>>>>
>>>> I registered myself.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> easy.selfip.com
>>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>
>>>> successfully activated.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d>
>>>>
>>>>
>>>>
>>>> Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a>
>>>>
>>>>
>>>>
>>>> Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a>
>>>>
>>>>
>>>>
>>>> Last Updated
>>>> <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com
>>>>
>>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline
>>>>
>>>>
>>>>
>>>> 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM
>>>>
>>>>
>>>> What next ???
>>>>
>>>>
>>>> On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote:
>>>>
>>>>> I dont have a Static IP.
>>>>>
>>>>> How can I ask IPKall to send call to my Asterisk ?
>>>>>
>>>>>
>>>>
>>>
>>
>
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_
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Re: [asterisk-users] Static IP
So, this will change :
register => 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure=very
fromdomain=proxy.ideasip.com
exten => _1101XXX,1,SetCallerID(Your Name <11012012600>)
exten =>
_1101XXX,2,Dial(SIP/${ext...@proxy.ideasip.com
)
exten => _1101XXX,3,Hangup
Feb 17 20:42:17 NOTICE[2746]: chan_sip.c:5529 sip_reg_timeout:--
Registration for '11012012...@proxy.ideasip.com' timed out, trying again
(Attempt #119)
-- parse_srv: SRV mapped to host proxy.ideasip.com, port 5060
-- Got SIP response 479 "Please don't use private IP addresses" back
from 208.97.25.11
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC wrote:
>
> hmmm Ok..
>
> Is this a Asterisk Question ?
>
> I have a setting as :
>
> Global Settings :
> ---
> SipIpkall = SIP/fwd
>
> Dialplan Entry :
>
> exten => 11012012600,1,Ringing call ringing
> exten => 11012012600,2,Wait(1) Wait 1 second for CID delivery from PRI
> exten => 11012012600,3,Answer Answer the line
> exten =>
> 11012012600,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-11012012600-Closer-park--999-1-TESTCAMP)
> exten => 11012012600,5,Hangup
>
>
> Registration String: :
> ---
> register
> =>11012012600:passw...@66.54.140.46<11012012600%3apassw...@66.54.140.46>
>
>
> Sip Entry :
> --
> [fwd]
> type=friend
> secret=password
> username=11012012600
> host=66.54.140.46
>
>
>
> I get in CLI ::
>
> Feb 17 19:03:17 NOTICE[2554]: chan_sip.c:5529 sip_reg_timeout:--
> Registration for '11012012...@66.54.140.46' timed out, trying again
> (Attempt #18)
>
> When I try to Ping from my CentOS , I can ping 66.54.140.46.
>
>
> On Thu, Feb 18, 2010 at 3:11 AM, David @ULC wrote:
>
>>
>> Looks like IdeaSip need STATIC ip else it doesnt work.
>>
>> .
>>
>>
>> On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote:
>>
>>> Ok
>>>
>>> I can use
>>>
>>> Dyndns.org
>>>
>>> I registered myself.
>>>
>>>
>>>
>>>
>>> easy.selfip.com
>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>
>>> successfully activated.
>>>
>>>
>>>
>>>
>>>
>>>
>>> Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d>
>>>
>>>
>>> Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a>
>>>
>>>
>>> Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a>
>>>
>>>
>>> Last Updated
>>> <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com
>>>
>>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline
>>>
>>>
>>> 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM
>>>
>>>
>>> What next ???
>>>
>>>
>>> On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote:
>>>
>>>> I dont have a Static IP.
>>>>
>>>> How can I ask IPKall to send call to my Asterisk ?
>>>>
>>>>
>>>
>>
>
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_
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Re: [asterisk-users] Static IP
Feb 17 19:19:04 NOTICE[2554]: chan_sip.c:10077 handle_response_peerpoke: Peer '11012012600' is now TOO LAGGED! (2567ms / 2000ms) On Thu, Feb 18, 2010 at 5:34 AM, David @ULC wrote: > > hmmm Ok.. > > Is this a Asterisk Question ? > > I have a setting as : > > Global Settings : > --- > SipIpkall = SIP/fwd > > Dialplan Entry : > > exten => 11012012600,1,Ringing call ringing > exten => 11012012600,2,Wait(1) Wait 1 second for CID delivery from PRI > exten => 11012012600,3,Answer Answer the line > exten => > 11012012600,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-11012012600-Closer-park--999-1-TESTCAMP) > exten => 11012012600,5,Hangup > > > Registration String: : > --- > register > =>11012012600:passw...@66.54.140.46<11012012600%3apassw...@66.54.140.46> > > > Sip Entry : > -- > [fwd] > type=friend > secret=password > username=11012012600 > host=66.54.140.46 > > > > I get in CLI :: > > Feb 17 19:03:17 NOTICE[2554]: chan_sip.c:5529 sip_reg_timeout:-- > Registration for '11012012...@66.54.140.46' timed out, trying again > (Attempt #18) > > When I try to Ping from my CentOS , I can ping 66.54.140.46. > > > On Thu, Feb 18, 2010 at 3:11 AM, David @ULC wrote: > >> >> Looks like IdeaSip need STATIC ip else it doesnt work. >> >> . >> >> >> On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote: >> >>> Ok >>> >>> I can use >>> >>> Dyndns.org >>> >>> I registered myself. >>> >>> >>> >>> >>> easy.selfip.com >>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com> >>> successfully activated. >>> >>> >>> >>> >>> >>> >>> Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d> >>> >>> >>> Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a> >>> >>> >>> Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a> >>> >>> >>> Last Updated >>> <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com >>> >>> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline >>> >>> >>> 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM >>> >>> >>> What next ??? >>> >>> >>> On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote: >>> >>>> I dont have a Static IP. >>>> >>>> How can I ask IPKall to send call to my Asterisk ? >>>> >>>> >>> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static IP
hmmm Ok.. Is this a Asterisk Question ? I have a setting as : Global Settings : --- SipIpkall = SIP/fwd Dialplan Entry : exten => 11012012600,1,Ringing call ringing exten => 11012012600,2,Wait(1) Wait 1 second for CID delivery from PRI exten => 11012012600,3,Answer Answer the line exten => 11012012600,4,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-LB-SALESLINE-11012012600-Closer-park--999-1-TESTCAMP) exten => 11012012600,5,Hangup Registration String: : --- register =>11012012600:passw...@66.54.140.46<11012012600%3apassw...@66.54.140.46> Sip Entry : -- [fwd] type=friend secret=password username=11012012600 host=66.54.140.46 I get in CLI :: Feb 17 19:03:17 NOTICE[2554]: chan_sip.c:5529 sip_reg_timeout:-- Registration for '11012012...@66.54.140.46' timed out, trying again (Attempt #18) When I try to Ping from my CentOS , I can ping 66.54.140.46. On Thu, Feb 18, 2010 at 3:11 AM, David @ULC wrote: > > Looks like IdeaSip need STATIC ip else it doesnt work. > > . > > > On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote: > >> Ok >> >> I can use >> >> Dyndns.org >> >> I registered myself. >> >> >> >> easy.selfip.com >> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com> >> successfully activated. >> >> >> >> >> Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d> >> >> Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a> >> >> Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a> >> >> Last Updated >> <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com >> >> <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline >> >> 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM >> >> >> What next ??? >> >> >> On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote: >> >>> I dont have a Static IP. >>> >>> How can I ask IPKall to send call to my Asterisk ? >>> >>> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static IP
Looks like IdeaSip need STATIC ip else it doesnt work. . On Thu, Feb 18, 2010 at 3:02 AM, David @ULC wrote: > Ok > > I can use > > Dyndns.org > > I registered myself. > > > easy.selfip.com > <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com> > successfully activated. > > > Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d> > Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a> > Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a> > Last Updated > <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com > <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline > 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM > > > What next ??? > > > On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote: > >> I dont have a Static IP. >> >> How can I ask IPKall to send call to my Asterisk ? >> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static IP
Ok I can use Dyndns.org I registered myself. easy.selfip.com <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com> successfully activated. Hostname <https://www.dyndns.com/account/services/hosts/?field=fqdn&sort=d>Service <https://www.dyndns.com/account/services/hosts/?field=redir&sort=a>Details <https://www.dyndns.com/account/services/hosts/?field=ip&sort=a>Last Updated <https://www.dyndns.com/account/services/hosts/?field=delta&sort=a>easy.selfip.com <https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com>Offline 127.0.0.1 (not in DNS)Feb. 17, 2010 4:30 PM What next ??????? On Thu, Feb 18, 2010 at 2:45 AM, David @ULC wrote: > I dont have a Static IP. > > How can I ask IPKall to send call to my Asterisk ? > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Static IP
I dont have a Static IP. How can I ask IPKall to send call to my Asterisk ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ideasip
I use IdeaSip with IPKall. How may channels are open when we use IdeaSip ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] Maximum call handling capacity on single server
On Mon, Feb 15, 2010 at 11:05 AM, Amit Patkar | Avhan Technologies Pvt. Ltd. wrote: > I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for > PSTN-IP gateway. What is the maximum call handling capacity I can achieve > with this server? You can handle a lot of pure sip calls. You don't say anything about the PCI / PCI-E bus on that machine, and purely speaking, nobody here knows whether that server can even physically terminate several cards. > I want at least 480 concurrent PSTN-IP calls. That mean I will have to > install minimum 4 x 4E1 cards and run 480 G.711 RTP sessions. No call > recording. No IVR. Pure gateway functionality. Can I achieve this capacity > with given server configuration? > If not, what kind of server is required to achieve this capacity. > > Has anyone done this? Please share results. If anybody has done this, they would run into the problems the others mentioned. If you can afford the phone bills you're going to incur with this setup, you're also going to be able to afford at least making 2x the capacity you really need for redundancy and business continuity purposes. Also, you're math is bad. If you're talking US, T1/PRI, you're only terminating 23 channels per T1/PRI, so if you need 480 channels, you're going to need 21 PRI's meaning 6x 4-port cards. At which point, you should seriously consider buying a hardware appliance like a Cisco 3845, and really you should buy two and split the lines over those two. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi debug in Asterisk 1.6?
On Sat, Feb 13, 2010 at 10:50 PM, Alejandro Recarey wrote: > Much to my surprise I tried to debug an AGI script today with "agi > debug" on the Asterisk CLI and it did not work. Plus, I could find no > reference on lie of it being removed. > > Is there another name for that command? I scanned the CLI help but > found nothing similar. Both my 1.6 boxes do not have the command but > my 1.4 box does. try agi set debug on In general, you should know that if asterisk CLI has command completion, meaning you can type 'agi' [tab] and it will give you your possible completions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know what would be killer?
On Thu, Feb 4, 2010 at 7:39 PM, Lyle Underwood wrote: > If call recordings were stored in stereo and the callers were evenly > distributed along the stereo spectrum. BAM. Cisco has this. It's called telepresence. It costs a LOT of money, and takes a LOT of bandwidth, but you do get spatial distribution with both video and audio. It requires multiple cameras, multiple monitors, multiple microphones, multiple speakers, but it does work. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
On Tue, Feb 9, 2010 at 10:26 AM, Klaus Darilion wrote: > Answering myself: muting means that the participants voice is ignored. Thank you for updating the wiki and the list. I looked into this when I was having problems with early 1.6.0.* MeetMe(), specifically the talker detection problem where beginnings and endings of sentences would be clipped and not mixed (exacerbated by SIP vad). I was able to tune MeetMe() and SIP better and solve my problem, but ConfBridge() certainly seemed promising. A big thanks to Josh Colp and others for making this happen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion wrote: > I wonder what "mute" should mean. Does it mean that the participant will > not receive any media, or that media sent by the participant will be > ignored, or both? Please post your discoveries to: http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge so we can all learn at together. I wrote that up when I couldn't find documentation on my own. Obviously it's short on details. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue with strategy=linear
Hi, Using asterisk 1.6.2.0 I have a queue definition with "strategy=linear". How do I jump to the next dialplan item after having tried (unsuccessfully) all queue members? If I use Queue(test,n) then only the first member is contacted. And if I omit the "n" option then all members are retried indefinitely. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd wrote: > Hello there, > > I'm trying to figure out how to run a PHP script on a remote machine and > still have access to the audio stream associated with the call. > > Ideally, I'd love to play/record audio files directly from/to the remote > server without having to copy them back and forth to the Asterisk > server. What is the best way to do this? > > Is it possible to combine EAGI with FastAGI in PHP? You don't specify how often / what proportion of the recordings need to be on a remote machine versus on the asterisk server. So you have two main things going on: 1) recordings, with a side order of distributing those to another machine 2) remote shell scripting First, the recordings can be done directly on a channel where the call is taking place. If this is one call, that's not so bad, but there get to be I/O contention issues when you try to record 'a lot' of calls simultaneously. Some people endorse working around that by writing recordings to a ramdisk, and then occasionally flushing those off to a real hard disk. You may prefer an alternate approach, which is that taken by commercial recording solutions. Oreka (which can be grabbed from sourceforge), and pretty much every commercial voip recording solution I've investigated, works by having you use libpcap (used in Ethereal/Wireshark) to watch ethernet device(s) where voip calls are taking place, grab the SIP headers that set up the RTP stream, and then write those recordings to disk on a dedicated recordings server. This requires explicit ethernet support by doing things like port mirroring, or using an old-school hub, etc. This has an advantage for you of providing a way to do recording directly on a machine that is NOT the asterisk server. No copying required as the recording is already where you want it. Second, the remote shell isn't so hard. ssh with keys, problem solved. You can do that directly from the asterisk dialplan using the System() command. This let's you tie the remote shell directly to a given call, where you can tune arguments accordingly. Of course, you can also do #1 with scripting and remote shell, or rsync with keys. If you don't need 'a lot' of simultaneous channels recorded, this may be more straightforward. You only have to learn asterisk, rather than asterisk and Oreka. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ongoing calls interface
I'm currently working on a PHP web interface to show (1) the registered endpoints and (2) their status: available, outgoing call or incoming call. (In the future, this interface should also be able to redirect calls etc.) (1) The interface already shows a list of all registered endpoints. For this, I had to work around Asterisk's delay in monitoring when an endpoint is unregistered. What I the interface does, is ping each endpoint to make sure it is still registered. Is there a better way to retrieve the endpoints still registered, instead of those that were registered an hour or so ago but might be unregistered already? (2) For each registered endpoint, the status needs to be displayed in the web interface. As I see it, there are two ways to retrieve this information. (a) Either I pull the information from Asterisk, by using some kind of "core show channels" command. I tried to use CDR to write calls made by the endpoints to a database. This works, but the rows are only written when a call is disconnected; I want to be able to see the status before that. (b) Another way is to keep track of notifications sent by endpoints. This is what I'm doing now. I added "action URIs" to all phones to make them send their data to a URL when they initiate a call, end a call etc. At that URL, a PHP script reads the data and saves it to a database. This works, but it feels like writing functionality that might already be available in Asterisk itself. Are there any other (better ways)? Which of the methods would be fastest for a large number of endpoints, and most reliable? Any help is greatly appreciated. With kind regards, David de Boer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN on phones?
I've been waiting for this for years. Except that snom phones are crap -- I would really like to see openvpn or ssh tunneling hacked into a Cisco phone... But it's still awesome. -dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, February 04, 2010 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OpenVPN on phones? --[ UxBoD ]-- wrote: > > Just taken delivery of a Snom 870 and one thing that did disappointment is > that you have to install a beta firmware to enable OpenVPN ... H ... > Please keep us informed on your thoughts about this phone. I'd like to know before buying one for testing. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 500 Internal Server Error on Cisco 7940 after INVITE
I have upgraded the phones to the most recent firmware (POS3-08-11-00) and * is "Version: 1:1.4.21.2~dfsg-3+lenny1" (debian). That doesn't look like cisco firmware to me... Unless I'm mistaken. What version are the phones on? (Settings => Status => Firmware Versions) -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with recordings of call using Monitor
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog wrote: > I'm using the default Asterisk function Monitor to record calls, but i have > some issue's with this, the problem is when a call is finished, it never mix > in & out together, bellow you can see my call configuration: Perhaps you would prefer to use MixMonitor() rather than Monitor() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium fax - sending fax call file vs manager originate
On Fri, Jan 29, 2010 at 3:09 PM, Hristo Benev wrote: > If I use call file with spool > Fax is send but if I use manager > I get > Any suggestions? Well, one obvious solution is to just use call file. Problem solved. Try changing your call manager setup to use a Local channel instead, and set up a context that does the dial within that context. That should give you better introspection into where things are failing / what you're doing wrong. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
However, if you're going to be doing massive joins for reporting, you're better off using something else (or running individual MySQL slaves, whose purpose is to run those complex queries and doing nothing else). In a past life, our MySQL database ran circles around Oracle, Informix, and DB2... until someone ran a massive join on the same server, which caused MySQL to crawl. Good distinction to make. I should have been more clear. I believe mysql has read only slave capabilities within a clustered environment, so your point about the slaves isn't out of the question. However I don't believe in database engines doing really anything other than transaction processing. That's why IMHO there should always be a distinction between the database backend and whatever software you're using to generate OLAP data (this software should NOT be the database engine). I know this is not a common opinion, but if we keep the database engine doing what it's good at and leave any report processing to external software, we're generally able to get better performance out of each individual piece... -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. This statement is absolute BS. Give me some factual, backed statements by trained database professionals who don't work for Microsoft or Oracle (OK, Sun) and we can talk. I guarantee that mysql can be made at least as fast as Oracle with a relational database that's designed, indexed and implemented properly. The problem with the backend is NEARLY ALWAYS a problem with the DBA. I hate to hear crap DBAs blame their problems on the backend. MySQL is top-notch and production ready if you are logical about your DB design. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
This is WAY OT but I had no idea what fnal.gov was, so I checked it out: http://computing.fnal.gov/xms/Services/Getting_Services/Web_at_Fermilab/Professional_Home_Pages_at_Fermilab And I quote "...professional information about themself..." About themself? Really? Really? That is all. Cheers Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Wednesday, January 27, 2010 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MYSQL problem On Wednesday 27 January 2010 01:34:57 Zhang Shukun wrote: > 2010/1/23 Steve Edwards : > > On Fri, 22 Jan 2010, Zhang Shukun wrote: > >> as you know, we can use MYSQL command to visit mysql database > >> > >> but if i use other database like Oracke,sybase,etc, Could i use MYSQL > >> command ? > > > > ODBC will do what you want. > > Thanks, while i think because oracle has no offical ODBC for linux system. I've used this driver with Oracle before with good results. Just make sure that you use the full Oracle libraries, not the Instant Client. The Instant Client has a severe resource leak that makes it inappropriate for long-running processes. http://home.fnal.gov/~dbox/oracle/odbc/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFAX and SendFAX questions
On Sun, Jan 24, 2010 at 2:51 AM, Magnus Benngård wrote: > Morning, > > Have some questions regarding receiving and sending faxes... > 1:st example: > exten => 101,1,Answer() > exten => 101,2,Wait(3) > exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) You example is correct in theory. The problem is that after receive fax completes, the sending side usually hangs up, so it's far safer to do your conversion and any other steps in the hangup context, as in h,1,System, etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Xfer issue between DataCenter and User Site
Admittedly I didn't read your SIP debug (on the mobile), but do you have reinvite=no set for the extensions and SIP trunks (providers)? This sounds on the surface like a classic case of the Mondays. Erm reinvites I mean. 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. 7. MoH stops, 8. Office user gets no audio 9. Incoming call is silent, and then call is dropped 10. Office user gets fed up of saying ‘hello??!?’ and hangs up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with picking out a digium card.
Some rack-mount servers I've encountered have an option to have the older-style PCI slots available in at least some slots. If you're really just using four FXS/FXO ports, it's unlikely you need very much horsepower, and you could use an older system for the foreseeable future. If you really need FXS/FXO, but want new non-PCI hardware, you might be better off considering an asterisk appliance that would convert FXS/FXO to SIP and let your new gear do the SIP, or just configure asterisk directly on that appliance. You would probably save power consumption versus a new server or even the old server currently in use. On Sun, Jan 17, 2010 at 3:25 PM, shawn bright wrote: > Hey all, > > We have been using a TDM400 card at work to provide our IVR. > We we have upgraded our server and now require the same capability, but on a > card that goes into a PCI Express. > Any suggestions would be greatly appreciated. > > oh, and it has to work with the zaptel drivers for linux. > > thanks all. > > sk > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 3:15 PM, Bruce Nik wrote: > Provided there is no comprehensive install guides (or is there?) yes I would > like to see an easy install script which can install it all. tar xvzf ./configure make (optional, do a 'make menuconfig') make install But the problem is that there are steps before the configure you need if you want support for more than barebones asterisk. Nobody knows what you personally need except you. Maybe I'm the only one who doesn't think it's so bad to build from source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik wrote: > Hi Guys, > Other than than yum repository (which fails when installing freepbx with it) > are there any automated install scripts out there that would install > Asterisk 1.6 or 1.4 onto a CentOS LAMP system? > If the script install FreePBX that would be a BONUS. > Thanks, > Bruce Do you like 'kitchen sink' installs? I can't think of any way to decide on an asterisk configuration that out-of-the-box would be right for everybody... as in, fax support? g729 licenses? whether or not to build against DAHDI? You get the idea. The only way I can think to do it would to be to build in a lot of stuff that most people would never want in their asterisk, which would then result in having to restart asterisk because you need a software update to a package that is a dependency for a part of asterisk you don't use anyway. Anybody who was using asterisk in a serious production environment would probably prefer the control of having most of what they don't want compiled out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
On Fri, Jan 15, 2010 at 1:54 AM, randall wrote: > does anybody know of another solution to this or is my conclusion above > simply all the choice there is? So let me get this straight. You're planning on buying multiple Gigabit, PoE switches, and you're quibbling over the price of running parallel data cable? The gigabit PoE switches are not cheap, at least if you're buying enterprise switches that actually deliver real gigabit, with full cross-sectional bandwidth. The cable isn't very much money, and if you double-wire now, you're ready when you have twice as many employees in the same space. Next, you don't say what this office is like, but I'm going to let you in on a little secret. Most people in an office rarely spike to a full 100Mbit connection. Do some bandwidth monitoring on your network and you'll discover that. A gigabit ethernet phone is a nice thing to have, but it's more a marketing thing than an actual necessity. Anybody that can afford a gigabit ethernet switching phone and true gigabit ethernet PoE backend can afford a second wire to every desk. Please let me know the use case if you find people can't be happy with a 100Mbit connection for the typical Windoze office environment. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to strip + from the caller-ID
Are you actually trying to strip off the + or are you doing it as part of
trying to check the callerid number to see if it is valid. If the later,
then consider REGEX()... here is a snippet from my privacy manager script...
; First lookup number in asterisk DB for a Caller ID name. exten =
s,n,Set(CALLERID(name)=${IF(${DB_EXISTS(cidname/${CALLERID(num)})}?${DB_RESULT}:${CALLERID(name)})})
; Now check against whitelist. exten =
s,n,GotoIf(${DB_EXISTS(whitelist/${CALLERID(num)})}?onwhitelist) ; Not on
whitelist, check it against blacklist exten =
s,n,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?onblacklist) ; Not on
blacklist either, check Caller ID number for anonymous conditions... ; If
all zeros (with or without international dialing + sign) then caller is
anonymous. exten = s,n,GotoIf($[${REGEX("^[+]?0+$" ${CALLERID(num)})} =
1]?unknown) ; If Caller ID is not a number at least 4 digits long (with or
without + sign) ; then caller is assumed to be anonymous. Alphabetic Caller
ID "numbers" ; will therefore be considered anonymous. Common numbers for
anonymous callers ; are: "asterisk, unknown, anonymous, private,
unavailable" which may be upper, lower ; or mixed case. The regular
expression catches everything non-numeric. ; If you want to permit a
specific non-numeric Caller ID "number" add it to whitelist. exten =
s,n,GotoIf($[${REGEX("^[+]?[0-9]{4\,}$" ${CALLERID(num)})} != 1]?unknown) ;
Caller ID looks good.
On Thu, Jan 14, 2010 at 9:11 AM, Danny Nicholas wrote:
> I saw something like this in another answer, but here’s an example that
> should work (would on 1.4)
>
> exten => s/_+X.,1,Set(TMPNAME=${CALLERID(name)})
>
> exten => s/_+X.,n,Set(CLEANNAME=CUT(TMPNAME|\+|2))
>
> exten => s/_+X.,n,Set(CALLERID(name)=${CLEANNAME})
>
>
>
> in my installations ${X:1} is a hit or miss proposition; CUT is a Known
> quantity.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Szasz Szabolcs
> *Sent:* Thursday, January 14, 2010 3:55 AM
>
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] how to strip + from the caller-ID
>
>
>
> Hi,
>
> How can I strip + from the front of the caller ID?
> I have tried this:
> exten => s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1})
>
> But it is not working.
>
>
> Szasz Szabolcs
>
> --
> _
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Inserting a wait in a sip dial SOLVED (kind of)
Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? Well, due to the lack of responses, either I missed something obvious or nobody cares. I'm really hoping I didn't miss something obvious... :). In any event, I got curious of my own old question and hacked out a work around: 0. Assume your extension is dumped into context 'mycontext' 1. You dial an internal extension 2. * Dials an external number (presumably another PBX device) 3. When the remote device answers, both parties are dumped into the DTMFworkaround context 4. The called party has its DTMF mode set to inband so that the tones are played out loud 4.5. Meanwhile, the calling party is dumped into an empty meeting conference that is used soley to bridge these two legs 5. When the tones are done, the called party is dumped into the bridged conference. 6. When the caller hangs up, the conference boots the callee [dtmfworkaround] exten => 6534,1,Goto(dtmfworkaround|6536|1) exten => 6534,2,Goto(dtmfworkaround|6535|1) exten => 6535,1,Answer() exten => 6535,n,Wait(1) exten => 6535,n,SIPDTMFMode(inband) exten => 6535,n,SendDTMF(1234) exten => 6535,n,MeetMe(101|MFqx|1234) exten => 6536,1,Answer() exten => 6536,n,MeetMe(101|MFqxA|1234) [mycontext] exten => 658,1,Dial(SIP/486,15,rG(dtmfworkaround^6534^1)) -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
'w' is really only supported on channels where digit-by-digit dialing is the norm, which generally means analog trunks (or digital trunks using CAS signaling). Thanks Kevin, that's what I figured (though not quite so concisely)... Going foward, is there any way to programmatically inject DTMF tones into an already-bridged channel? So: 1. dial 12345 2. connect SIP provider to * extension 3. wait 2 seconds programmatically 3. inject 567 DTMF tones into channel to signal remote PBX of extension to dial I'm hoping there's another way to skin this cat. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
This doesn't work? Dial(SIP/*31#ww061234123412) When I was browsing the sip debugs, it seemed that the 'w' was not being honored for one reason or another. My thought at the time was maybe it didn't work at all over SIP. Does the w *just* work with dahdi or does it work over sip as well (assuming the provider honors it)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a wait in a sip dial
But then the other peer says: -- Called *31#w06123456...@xs4all-out -- SIP/xs4all-out-0234 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION' Anyone an idea where i should look, or how i should change it, so that i do get a wait before sending the rest of the number to the sip peer. I don't have an answer for this but am holding my breath that *someone* does. I ran into a similar situation (dial a number, then wait, then dial an extension via SIP to PSTN) a few weeks ago and never figured out a resolution... My THOUGHT is that you would have to manually inject the DTMF into the stream somehow after the SIP provider connects the call... Thanks Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use AGI php script function $agi -> exec_dial
You might find this helpful:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
Regards,
On Mon, Jan 11, 2010 at 2:19 AM, Zhang Shukun wrote:
> hi,
>
> i want to use $agi -> exec_dial() to dial .
>
> this is in extention.conf
>
> [tutorial]
> exten => 1234,1,Dial(SIP/ivan)
>
> is that i use
>
> $agi -> exec_dial("SIP","tutorial|1234|1")
>
> can dial ?
>
> BTW, i want to know some turorial on how to use PHPAGI funtions? can
> you tell me some?
>
> Thanks!
>
>
> --
> Best regards,
> Sucan
>
> --
> _
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>
--
David Cunningham
Voisonics Limited
IVR development, VOIP consultancy
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3411 5024
Australia: +61 (0) 2 9037 2180
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Re: [asterisk-users] Really Silly Question From Total Newbie
>> > Mike > >> > > >> > ___ > >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > > >> > asterisk-users mailing list > >> > To UNSUBSCRIBE or update options visit: > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > >> > ___ > >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > > >> > asterisk-users mailing list > >> > To UNSUBSCRIBE or update options visit: > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > >> > >> ___ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free FaxForAsterisk ReceiveFAX not working
On Fri, Jan 8, 2010 at 1:47 PM, Daniel Araujo wrote: >I have the same issue on my Asterisk installation (Asterisk 1.4.25). > As you can see, the T38 module isn't enabled on my installation. Tried ask > google how to make it work, but found no hints yet. > Anyone can help us? If you want T.38, you should be using 1.6. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
And how will we ever re-write the 10+-year-old RFCs which no longer hold relevance to modern email clients if nobody goes against the grain and does what makes sense rather than what has been generally accepted? -Dave And to add on to this: aside from whether you think it is silly or not, there are: 1) RFC's 2) List rules And when both of those tell you to bottom-post, then who are you to decide otherwise, just because you think it is silly? Well, maybe I think it is silly that I cannot hit you in the face everytime you say "I", would you allow me to hit you, or would you protest and demand I keep to the rules that tell me I can't do that? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I would have read your message but I couldn't find it amongst all of this garbage... :) -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne Sent: Friday, January 08, 2010 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please remove me from the mailing list. On 8 Jan 2010, at 16:03, Randy R wrote: > > On Fri, Jan 8, 2010 at 3:39 PM, Steve Howes wrote: >> On 8 Jan 2010, at 13:52, John Novack wrote: >>> Steve Howes wrote: On 8 Jan 2010, at 02:28, John Novack wrote: > Careful, or Steve will "un top post" YOU! I like it in the past. Leave me alone ;) >>> Different Steve!! >> >> I agree with him though :P >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > About what? :-| About dirty top-posters? W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
On Fri, Jan 8, 2010 at 8:59 AM, William Stillwell (Lists) wrote: > This is what I was using at the time: > > asterisk-1.4.21.2 I really, really prefer the faxing in 1.6. It's so nice to configure compared to 1.4. I'll leave it to the ChangeLog and anybody else who wants to chime in on actual differences. > I was using PSTN. Great. Because trying to track down voip faxing problems is much worse. > hardware provided determine from examining server logs. But some callers > just could not send a fax, it would fail every time, and I just couldn't > reproduce it.. Did you ever record your faxes? When I was troubleshooting things, I started recording 100% of faxes, and then just blowing them away after a few days with a cron. If I wanted to go back and troubleshoot a particular customer, I could filter by their calls and listen to what was going on. It was amazing how lousy some of the faxes were and it was obviously the customer's fault. I never would have been able to tell that without listening to the audio recordings of the fax transmission. In other cases, it was robodialers wardialing the world, and they weren't even sending a fax. I discovered I had to be VERY careful how I calculated error rate. If you count by absolute successes and failures, the early failure rate looked awful. This was directly correlated to the customers with crap connections retrying the same faxes that were never going to succeed over and over again. When I instead sorted successes and failures by sending phone number, I got very high 90s success rate. This of course, also requires that you're keeping logging in a way that makes this kind of diagnosis possible. Hopefully you have good records. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote: > problem I'm running into is if the DNS server is not responding, the > script hangs and waits for 30 seconds before returning to the Asterisk > dialplan. I would like a timeout of 1 second, then return. A few things... * stop using DNS? Problem solved. * put nagios monitoring on your DNS server? * put in a second DNS server, and tune your DNS timeout to a very low value in /etc/resolv.conf (read the man page) before jumping to next server? Or you could use the Perl language feature, which is called 'alarm'. Google around for some code samples. None of these are actually specific to asterisk, as it turns out. I don't know of any explicit asterisk method to force a timeout. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
