Re: [asterisk-users] Question about Registrations
One way of doing something when a peer registers is to use AMI to monitor events and when a register event occurs do what you want. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 23, 2011, at 7:08 PM, Alex Balashov wrote: > On 09/23/2011 09:59 PM, CDR wrote: > >> In Trunk, or earlier, is it possible to execute an AGI or any piece of >> the Diaplan when a new peer registers successfully? > > No. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Registrations
On 09/23/2011 09:59 PM, CDR wrote: In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? No. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Registrations
In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? Venefax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DTMF problem
On Sun, Sep 18, 2011 at 07:51:43PM -0400, Zeeshan A Zakaria wrote: > This DTMF problem has always been there and there is no real solution > for it, other than using those expensive PBX systems like that from > Avaya, Cisco, etc. This problem happens when you are sending inband > DTMF tones. Via softphone you are sending out-of-band DTMF which is > basically SIP messages. You can emulate this feature from the Expensive PBX system by setting: relaxdtmf=yes in the case of SIP, option may vary with Techology. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
Shaun Ruffell wrote: > On Fri, Sep 23, 2011 at 09:30:59AM -0400, cov...@ccs.covici.com wrote: > > Kevin P. Fleming wrote: > >> On 09/23/2011 02:50 AM, Ishfaq Malik wrote: > >>> On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: > > So am I correct in assuming dahdi_dummy isn't needed/useful > anymore? > >>> > >>> Application MeetMe will not work without it. > >> > >> This is completely incorrect. MeetMe never relied on dahdi_dummy > >> specifically, it requires DAHDI to have a working timing source. > >> Yes, at one point dahdi_dummy was available to provide a timing > >> source if there weren't any DAHDI cards in the system... but it > >> is no longer necessary. DAHDI is now able to provide timing and > >> audio mixing using kernel timers using a built-in timer, so there > >> is no need for a separate module. The ChangeLog entry above is > >> correct, as of DAHDI 2.4 and later. > > > > So, how do I get this to work -- when I tried to do this, I could > > get a conference all right, but it would not record the conference > > till I actually loaded dahdi-dummy -- which seems to be still > > built. I am using 9729 out of trunk. > > John, > > As kpfleming said, dahdi_dummy is no longer built by default. > Revision 9729 you referenced was first released in 2.5.0 which > definitely does not use dahdi_dummy by default. > > Perhaps you believe you were able to load dahdi_dummy because dahdi > is aliased to dahdi_dummy and "before" loading it you were using > confbridge? > > Below you can see how only dahdi is needed for timing and > conferencing since the timers are processed in the same function > that handles the conferencing: > > You can modprobe dahdi_dummy but only 'dahdi' is loaded and > dahdi_test will work fine... > > # modprobe dahdi_dummy > # lsmod | grep dahdi > dahdi 196680 0 > crc_ccitt 6337 1 dahdi > # dahdi_test -v -c 3 > Opened pseudo dahdi interface, measuring accuracy... > > 8192 samples in 8191.592 system clock sample intervals (99.995%) > 8192 samples in 8190.720 system clock sample intervals (99.984%) > 8192 samples in 8191.288 system clock sample intervals (99.991%) > --- Results after 3 passes --- > Best: 99.995 -- Worst: 99.984 -- Average: 99.990234, Difference: 99.990233 > > But you can do the same thing only by loading dahdi and not > dahdi_dummy... > > # modprobe -r dahdi > # lsmod | grep dahdi > # dahdi_test > Unable to open dahdi interface: No such file or directory > # modprobe dahdi > # dahdi_test -v -c2 > Opened pseudo dahdi interface, measuring accuracy... > > 8192 samples in 8199.624 system clock sample intervals (100.093%) > 8192 samples in 8182.688 system clock sample intervals (99.886%) > --- Results after 2 passes --- > Best: 99.907 -- Worst: 99.886 -- Average: 99.896633, Difference: 99.989697 > > DAHDI will still use the timing from an installed card if available, > but now it is smart enough to detect if there is not a card > installed or operating properly and still provide timing without > requiring the user to load "dahdi_dummy" explicitly. You are correct, when meetme didn't work, I did not even load dahdi at all -- that was the confusion. I am surprised that the modprobe of dahdi-dummy even succeeds, but I guess it does not matter. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls
Hi All;
I noticed in the queues.conf the configuration for recording the calls in the
queuing, and regarding to the filename (or any other parameter), it is written
that I can determine the filename using the command:
Set(MONITOR_FILENAME=foo)
But it should be called from the dialing plan, but really i did not understand
how to call it from the dialing plan.
Well, for example this is my dialing plan to route for the queuing, how I can
set the filename:
[CustomerSupport]
include => Internal
exten => s,1,Queue(CustomerSupport,t,,,120)
exten => s,2,Macro(voicemail,SIP/reception)
By the way, I need in the filename to appear the following:
The SIP username for the IP Phone that the call is routed for it
The calling number
The Time of the call
Actually for the outbound recording, I am using the below command (I mentioned
it to declare the time format I am using and to declare how the filename to be
named):
exten =>
_9Z,1,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
So I hope if someone can help me to write the Set(MONITOR_FILENAME=foo) in a
way to acheive same format the filename of the recorded outgoing calls (in
addition that until now I am not able to know where I have to place the
Set(MONITOR_FILENAME=foo).
For example, should I place it as following:
exten => s,1,Set(MONITOR_FILENAME=.)
exten => s,2,Queue(CustomerSupport,t,,,120)
exten => s,3,Macro(voicemail,SIP/reception)
Appreciate if someone help me plz.
Regards
Bilal
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Re: [asterisk-users] dahdi_dummy required?
On Fri, Sep 23, 2011 at 09:30:59AM -0400, cov...@ccs.covici.com wrote: > Kevin P. Fleming wrote: >> On 09/23/2011 02:50 AM, Ishfaq Malik wrote: >>> On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: So am I correct in assuming dahdi_dummy isn't needed/useful anymore? >>> >>> Application MeetMe will not work without it. >> >> This is completely incorrect. MeetMe never relied on dahdi_dummy >> specifically, it requires DAHDI to have a working timing source. >> Yes, at one point dahdi_dummy was available to provide a timing >> source if there weren't any DAHDI cards in the system... but it >> is no longer necessary. DAHDI is now able to provide timing and >> audio mixing using kernel timers using a built-in timer, so there >> is no need for a separate module. The ChangeLog entry above is >> correct, as of DAHDI 2.4 and later. > > So, how do I get this to work -- when I tried to do this, I could > get a conference all right, but it would not record the conference > till I actually loaded dahdi-dummy -- which seems to be still > built. I am using 9729 out of trunk. John, As kpfleming said, dahdi_dummy is no longer built by default. Revision 9729 you referenced was first released in 2.5.0 which definitely does not use dahdi_dummy by default. Perhaps you believe you were able to load dahdi_dummy because dahdi is aliased to dahdi_dummy and "before" loading it you were using confbridge? Below you can see how only dahdi is needed for timing and conferencing since the timers are processed in the same function that handles the conferencing: You can modprobe dahdi_dummy but only 'dahdi' is loaded and dahdi_test will work fine... # modprobe dahdi_dummy # lsmod | grep dahdi dahdi 196680 0 crc_ccitt 6337 1 dahdi # dahdi_test -v -c 3 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8191.592 system clock sample intervals (99.995%) 8192 samples in 8190.720 system clock sample intervals (99.984%) 8192 samples in 8191.288 system clock sample intervals (99.991%) --- Results after 3 passes --- Best: 99.995 -- Worst: 99.984 -- Average: 99.990234, Difference: 99.990233 But you can do the same thing only by loading dahdi and not dahdi_dummy... # modprobe -r dahdi # lsmod | grep dahdi # dahdi_test Unable to open dahdi interface: No such file or directory # modprobe dahdi # dahdi_test -v -c2 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8199.624 system clock sample intervals (100.093%) 8192 samples in 8182.688 system clock sample intervals (99.886%) --- Results after 2 passes --- Best: 99.907 -- Worst: 99.886 -- Average: 99.896633, Difference: 99.989697 DAHDI will still use the timing from an installed card if available, but now it is smart enough to detect if there is not a card installed or operating properly and still provide timing without requiring the user to load "dahdi_dummy" explicitly. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] looking for free DID 708-839
Are there any free DID in Illinois 708-839 or area? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
On 09/23/2011 12:16 PM, Mehmet Avcioglu wrote: On Sep 23, 2011, at 8:07 PM, Danny Nicholas wrote: Just a WAG - 4 is the error level returned by your php script, where it normally returns 0. Yes would thing so. But at no place in my script I intentionally exit with 4. I believe 4 is SIGILL (Illegal Instruction) so my script might be seg faulting somewhere? Should I be going after this? It is a php script and php doesn't log anything for these instances. No, 4 isn't SIGILL; result codes generated by uncaptured signals are always negative, I believe. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Postgresql Reconnect on connection failure
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Hiller Sent: Friday, September 23, 2011 1:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Postgresql Reconnect on connection failure Currently if asterisk loses its connection to the postgresql it does not attempt to reconnect. I have searched all over for a setting that would have asterisk attempt to reconnect but I can not find anything. Is there something I am missing? Thanks! -Eric So you like pain, huh? Have you read this article? http://climbing-the-hill.blogspot.com/2008/04/asterisk-realtime-architecture .html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.7.0 Now Available
The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have gone into features.c in this release (call parking, built-in transfers, call pickup, etc.). NOTE: Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as a result of Google's acquisition of GIPS, who produced (and provided licenses for) the iLBC codec. If you are a user of Asterisk and iLBC together, and you've already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking). More information is available on the Asterisk blog: http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ The following is a sample of the issues resolved in this release: * Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html * Significant fixes and improvements to parking lots. (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) * Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to. A change to Asterisk adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly others. (In essence, this change should make res_timing_timerfd usable.) * Resolve segfault when publishing device states via XMPP and not connected. (Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested by Jonathan Rose) * Refresh peer address if DNS unavailable at peer creation. (Closes issue ASTERISK-18000) * Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett) * Remove unnecessary libpri dependency checks in the configure script. (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett) * Update get_ilbc_source.sh script to work again. (Closes issue ASTERISK-18412) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Postgresql Reconnect on connection failure
Currently if asterisk loses its connection to the postgresql it does not attempt to reconnect. I have searched all over for a setting that would have asterisk attempt to reconnect but I can not find anything. Is there something I am missing? Thanks! -Eric -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
On Sep 23, 2011, at 8:07 PM, Danny Nicholas wrote: > Just a WAG - 4 is the error level returned by your php script, where it > normally returns 0. Yes would thing so. But at no place in my script I intentionally exit with 4. I believe 4 is SIGILL (Illegal Instruction) so my script might be seg faulting somewhere? Should I be going after this? It is a php script and php doesn't log anything for these instances. Thanks -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Force a SIP friend to use a certain IP?
So I was hoping I would be able to set the source IP that we use when talking to the two different SIP friends. I see externip in general options, but is there nothing equivalent that can be set per user/peer? Hi, as far as I know, you cant do this on a per peer basis. I suppose you run two asterisk daemons, each one of them on a different external IP. In this setup you can route calls from A over one asterisk daemon and calls from B over the other asterisk daemon. Sounds a little bit like an overkill scenario, but it woul work. best regards, Ruben Thanks, I was afraid of that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Problem
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mehmet Avcioglu Sent: Friday, September 23, 2011 12:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AGI Problem Hello, I have an AGI script that occasionally disappears without completing its action and asterisk logs the following. AGI Script script.php completed, returning 4 Spawn extension (context, 0123456, 2) exited non-zero on 'Local/0123456@context-f46e;1' I figured this was due to channel hanging up and * sending a SIGHUP to the script and added a catch and ignore for SIGHUP and SIGPIPE. But I still have instances where AGI script gets lost. I am running 1.8. Any ideas what "returning 4" really means, where should I concentrate? Thanks -- Mehmet Just a WAG - 4 is the error level returned by your php script, where it normally returns 0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] AGI Problem
Hello, I have an AGI script that occasionally disappears without completing its action and asterisk logs the following. AGI Script script.php completed, returning 4 Spawn extension (context, 0123456, 2) exited non-zero on 'Local/0123456@context-f46e;1' I figured this was due to channel hanging up and * sending a SIGHUP to the script and added a catch and ignore for SIGHUP and SIGPIPE. But I still have instances where AGI script gets lost. I am running 1.8. Any ideas what "returning 4" really means, where should I concentrate? Thanks -- Mehmet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Force a SIP friend to use a certain IP?
> So I was hoping I would be able to set the source IP that we use when > talking to the two different SIP friends. I see externip in general > options, but is there nothing equivalent that can be set per user/peer? Hi, as far as I know, you cant do this on a per peer basis. I suppose you run two asterisk daemons, each one of them on a different external IP. In this setup you can route calls from A over one asterisk daemon and calls from B over the other asterisk daemon. Sounds a little bit like an overkill scenario, but it woul work. best regards, Ruben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] usb hubs bluetooth chan_mobile
Hello, I have a question regarding a usb hub which is connected with a usb bluetooth adapter I setup asterisk16 with chan_mobile.Is working good. 1). When I use the bluetooth adapter into computer usb port is working voice 2 ways without delay : test OK 2). When I use the bluetooth adapter into usbhub (usbsplitter) which is connected to the usb port of computer the voice is one way : test FAILED Is some problem that the usbhubs don't support a2db or audio channels , only stick drive and printers? Do you have a solution for me? Thank in adavanced Sincerely Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Force a SIP friend to use a certain IP?
Suppose I have two IP aliases on one asterisk box. I have to talk to SIP friend "A" using IP x.x.x.x and I have to talk to SIP friend "B" using IP y.y.y.y. (In case you're wondering, the reason is that we have two accounts with a service provider and different features and rates are tied to the two different accounts.) So I was hoping I would be able to set the source IP that we use when talking to the two different SIP friends. I see externip in general options, but is there nothing equivalent that can be set per user/peer? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] sending fax using chan_capi
Hi,
I tried to sendfax a text file, it was received successfully and the context
were in ascii format (readable form). As I tried to send a fax in .tiff
format (converted from pdf format using ghostscript), the context I received
in fax is in binary form. The dial plan is listed below;
exten => 100,1,Verbose(> Sending Dialogic Diva Fax...)
exten => 100,n,set(BeforeFaxTime=${EPOCH})
exten => 100,n,capicommand(sendfax,/tmp/out.tiff,732-XXX-,Dialogic Diva
Test Sendfax)
exten => 100,n,HangUp()
exten => h,1,set(ElapsedFaxTime=$[${EPOCH}-${BeforeFaxTime}])
exten =>
h,n,AGI(printfaxresults.sh,${FAXSTATUS},${FAXREASON},${FAXREASONTEXT},${FAXRATE},${FAXRESOLUTION},${FAXFORMAT},${FAXCFFFORMAT},${FAXPAGES},${FAXID},${FAXEXTEN},${ElapsedFaxTime},FaxesSent.log)
Please advice, how can I send fax in image format using T.30
--
Regards,
Ahmed Munir Chohan
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Re: [asterisk-users] Digium ISDN card
On Fri, Sep 23, 2011 at 06:31:16PM +0530, michael k wrote: > Can anybody tell me which pci or pci express digium card can be used > to connect my asterisk server and the ISDN pri line with 30 channels ? You need to search for an E1 card (32 channels total, 30 voice): http://www.digium.com/en/products/digital/single-span/ Exact model depends on type of PCI interface. Don't forget to set the jumper from T1 to E1 :) -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with multiple sip-peers against the same host
Dear List, Thank you for your suggestions. This turned out to be an issue with our SIP provider, and has now been resolved. Regards, David On 2011-09-23 14:44, David Björkevik wrote: > Leandro, > > Thank you for your input! > > I tried this and it's still the same. > (although I still have _unrelated_ peers with the insecure entry) > > /David > > On 2011-09-23 14:24, Leandro Dardini wrote: >> Add "match_auth_username=yes" in the [general] section of sip.conf >> >> Remove from each peer any "insecure" entry >> >> Usually I add also "auth", "defaultuser" and "username" to the peer >> definition, but some of these entries are not needed. >> >> Leandro >> >> 2011/9/23 David Björkevik mailto:da...@dynamore.se>> >> >> Dear list, >> >> We are switching to a new provider for SIP-trunks. We have 20 numbers, >> each defined as a separate SIP peer. >> >> With the old provider everything works. >> >> When switching to the new provider's account data, it only works as long >> as I only define one of the accounts. If multiple accounts are defined, >> I can only place outgoing calls on one of them, for the other(s) >> authentication fails, "FORBIDDEN". >> >> It is almost like Asterisk is using just one of the defined passwords to >> authenticate all peers on that host. >> >> Any input is very appreciated. >> >> Regards >> David Björkevik, Engineer >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every 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 -- David Björkevik, Engineer DYNAmore Nordic AB - http://www.dynamore.se/ Full contact information: http://people.dynamore.se/david Voice: +46 (0)13-23 66 80 On July 1, DYNAmore Nordic AB acquired all of the business of Engineering Research. Read more on www.dynamore.se/dynamore-purchase Note the new @dynamore.se E-mail endings, previous @erab.se endings will work until the end of 2011. signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
Kevin P. Fleming wrote: > On 09/23/2011 02:50 AM, Ishfaq Malik wrote: > > On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: > >> I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been > >> reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy > >> should be used to provide an interface for Asterisk to get kernel > >> timing. - espescially if using timing-dependant modules. > >> > >> I have a minor question: is dahdi_dummy necessary or useful anymore - > >> espescially for users who don't have DAHDI hardware? > >> > >> I ask because I just checked out dahdi 2.5 from svn& built (against > >> the Linux kernel 3.0) > >> > >> I noticed that dahdi_dummy didn't seem to be built; when I poked around > >> in the changelog, I saw: > >> * README: README: Remove references to dahdi_dummy. Since > >>dahdi_dummy is no longer required remove the references from > >>README. (issue #17959) Reported by: glen201 Origin: > >>http://svnview.digium.com/svn/dahdi?view=rev&rev=9308 > >> > >> So am I correct in assuming dahdi_dummy isn't needed/useful anymore? > > > > Application MeetMe will not work without it. > > This is completely incorrect. MeetMe never relied on dahdi_dummy > specifically, it requires DAHDI to have a working timing source. Yes, > at one point dahdi_dummy was available to provide a timing source if > there weren't any DAHDI cards in the system... but it is no longer > necessary. DAHDI is now able to provide timing and audio mixing using > kernel timers using a built-in timer, so there is no need for a > separate module. The ChangeLog entry above is correct, as of DAHDI 2.4 > and later. So, how do I get this to work -- when I tried to do this, I could get a conference all right, but it would not record the conference till I actually loaded dahdi-dummy -- which seems to be still built. I am using 9729 out of trunk. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium ISDN card
compare the prices between sangoma and digium pri boards! Sangoma's oards here in Germany are cheaper as the ones from digium. if you need detailed help, you can contact me, and I can workout for you something as well as helping you setting up your pbx! Tamer Am 23.09.2011 15:01, schrieb michael k: > Hi All, > > I am new in asterisk. In my office we have purchased ISDN > pri line with 30 channels. we have more than 60 soft phone nodes and the > internal asterisk connectivity between extensions are working with soft > phones. Can anybody tell me which pci or pci express digium card can be > used to connect my asterisk server and the ISDN pri line with 30 > channels ? Please assist me to do if possible > > > > Michael.k > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Digium ISDN card
Hi All, I am new in asterisk. In my office we have purchased ISDN pri line with 30 channels. we have more than 60 soft phone nodes and the internal asterisk connectivity between extensions are working with soft phones. Can anybody tell me which pci or pci express digium card can be used to connect my asterisk server and the ISDN pri line with 30 channels ? Please assist me to do if possible Michael.k -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with multiple sip-peers against the same host
Please check no other peers with "insecure" entry are registered from the same IP. Asterisk takes some shortcut and try authenticating peers by IP address before authenticating them by username/password. Leandro 2011/9/23 David Björkevik > Leandro, > > Thank you for your input! > > I tried this and it's still the same. > (although I still have _unrelated_ peers with the insecure entry) > > /David > > On 2011-09-23 14:24, Leandro Dardini wrote: > > Add "match_auth_username=yes" in the [general] section of sip.conf > > > > Remove from each peer any "insecure" entry > > > > Usually I add also "auth", "defaultuser" and "username" to the peer > > definition, but some of these entries are not needed. > > > > Leandro > > > > 2011/9/23 David Björkevik mailto:da...@dynamore.se>> > > > > Dear list, > > > > We are switching to a new provider for SIP-trunks. We have 20 > numbers, > > each defined as a separate SIP peer. > > > > With the old provider everything works. > > > > When switching to the new provider's account data, it only works as > long > > as I only define one of the accounts. If multiple accounts are > defined, > > I can only place outgoing calls on one of them, for the other(s) > > authentication fails, "FORBIDDEN". > > > > It is almost like Asterisk is using just one of the defined passwords > to > > authenticate all peers on that host. > > > > Any input is very appreciated. > > > > Regards > > David Björkevik, Engineer > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com-- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > >http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > David Björkevik, Engineer > DYNAmore Nordic AB - http://www.dynamore.se/ > Full contact information: http://people.dynamore.se/david > Voice: +46 (0)13-23 66 80 > > On July 1, DYNAmore Nordic AB acquired all of the business of > Engineering Research. Read more on www.dynamore.se/dynamore-purchase > > Note the new @dynamore.se E-mail endings, previous > @erab.se endings will work until the end of 2011. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with multiple sip-peers against the same host
Leandro, Thank you for your input! I tried this and it's still the same. (although I still have _unrelated_ peers with the insecure entry) /David On 2011-09-23 14:24, Leandro Dardini wrote: > Add "match_auth_username=yes" in the [general] section of sip.conf > > Remove from each peer any "insecure" entry > > Usually I add also "auth", "defaultuser" and "username" to the peer > definition, but some of these entries are not needed. > > Leandro > > 2011/9/23 David Björkevik mailto:da...@dynamore.se>> > > Dear list, > > We are switching to a new provider for SIP-trunks. We have 20 numbers, > each defined as a separate SIP peer. > > With the old provider everything works. > > When switching to the new provider's account data, it only works as long > as I only define one of the accounts. If multiple accounts are defined, > I can only place outgoing calls on one of them, for the other(s) > authentication fails, "FORBIDDEN". > > It is almost like Asterisk is using just one of the defined passwords to > authenticate all peers on that host. > > Any input is very appreciated. > > Regards > David Björkevik, Engineer > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- David Björkevik, Engineer DYNAmore Nordic AB - http://www.dynamore.se/ Full contact information: http://people.dynamore.se/david Voice: +46 (0)13-23 66 80 On July 1, DYNAmore Nordic AB acquired all of the business of Engineering Research. Read more on www.dynamore.se/dynamore-purchase Note the new @dynamore.se E-mail endings, previous @erab.se endings will work until the end of 2011. signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Native bridging to SIP endpoints on the same NAT'd network
Hi, I have the following setup: Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the 2 endpoints directly, given that they are on the same non-routeable private net. With directmedia=nonat, I see Asterisk report the bridging of the calls but both sides of the call are routed to the originating endpoint so effectively, the call becomes an echo-loop. There is no audio on the second end-point although the call remains up. I assume this is some sort of firewall/nat/routing issue. Could someone explain what is possibly going on and perhaps offer a solution? Cheers, Richard. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with multiple sip-peers against the same host
Add "match_auth_username=yes" in the [general] section of sip.conf Remove from each peer any "insecure" entry Usually I add also "auth", "defaultuser" and "username" to the peer definition, but some of these entries are not needed. Leandro 2011/9/23 David Björkevik > Dear list, > > We are switching to a new provider for SIP-trunks. We have 20 numbers, > each defined as a separate SIP peer. > > With the old provider everything works. > > When switching to the new provider's account data, it only works as long > as I only define one of the accounts. If multiple accounts are defined, > I can only place outgoing calls on one of them, for the other(s) > authentication fails, "FORBIDDEN". > > It is almost like Asterisk is using just one of the defined passwords to > authenticate all peers on that host. > > Any input is very appreciated. > > Regards > David Björkevik, Engineer > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
On 09/23/2011 02:50 AM, Ishfaq Malik wrote: On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy should be used to provide an interface for Asterisk to get kernel timing. - espescially if using timing-dependant modules. I have a minor question: is dahdi_dummy necessary or useful anymore - espescially for users who don't have DAHDI hardware? I ask because I just checked out dahdi 2.5 from svn& built (against the Linux kernel 3.0) I noticed that dahdi_dummy didn't seem to be built; when I poked around in the changelog, I saw: * README: README: Remove references to dahdi_dummy. Since dahdi_dummy is no longer required remove the references from README. (issue #17959) Reported by: glen201 Origin: http://svnview.digium.com/svn/dahdi?view=rev&rev=9308 So am I correct in assuming dahdi_dummy isn't needed/useful anymore? Application MeetMe will not work without it. This is completely incorrect. MeetMe never relied on dahdi_dummy specifically, it requires DAHDI to have a working timing source. Yes, at one point dahdi_dummy was available to provide a timing source if there weren't any DAHDI cards in the system... but it is no longer necessary. DAHDI is now able to provide timing and audio mixing using kernel timers using a built-in timer, so there is no need for a separate module. The ChangeLog entry above is correct, as of DAHDI 2.4 and later. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 FXO stopped working
Hi list I have 2 servers with a TDM400 card, port 1 populated by an FXO (red) module), port 4 populated with an FXS module. I am using dahdi linux and tools 2.5.0.1. The servers are running CentOS 4 and the other box CentOS 6. Both modules have been working fine but recently stopped working, when i start dahdi with just FXS enabled everything is fine. This is the error i get : Loading DAHDI hardware modules: wctdm: [ OK ] Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span [FAILED] This is in my system.conf : fxoks=1 echocanceller=mg2,1 # channel 2, WCTDM/4/1, no module. # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 # Global data loadzone= nl defaultzone = nl When i run dahdi_genconf it doesn't detect the module either : # Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 23 11:24:16 2011 # Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) # channel 1, WCTDM/4/0, no module. # channel 2, WCTDM/4/1, no module. # channel 3, WCTDM/4/2, no module. fxsks=4 echocanceller=mg2,4 I already replaced both FXO modules with new ones but without result. Ideas anyone? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy required?
On Thu, 2011-09-22 at 22:19 -0600, Troy Telford wrote: > I'm running Asterisk 1.8.4.4; I'm new to asterisk, and have been > reading the Asterisk Definitive Guide, and it mentions that dahdi_dummy > should be used to provide an interface for Asterisk to get kernel > timing. - espescially if using timing-dependant modules. > > I have a minor question: is dahdi_dummy necessary or useful anymore - > espescially for users who don't have DAHDI hardware? > > I ask because I just checked out dahdi 2.5 from svn & built (against > the Linux kernel 3.0) > > I noticed that dahdi_dummy didn't seem to be built; when I poked around > in the changelog, I saw: > * README: README: Remove references to dahdi_dummy. Since > dahdi_dummy is no longer required remove the references from > README. (issue #17959) Reported by: glen201 Origin: > http://svnview.digium.com/svn/dahdi?view=rev&rev=9308 > > So am I correct in assuming dahdi_dummy isn't needed/useful anymore? Application MeetMe will not work without it. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
