Re: [asterisk-users] set global variable
Thanks for ideas, I thing I've found another way. It's possible to store data in database, alter them via manager and use in dialplan. Exactly, what I needed. Just hope, it will work Jan __ > Od: [EMAIL PROTECTED] > Komu: Asterisk Users Mailing List - Non-Commercial > Discussion > Datum: 13.09.2006 21:45 > Předmět: Re: [asterisk-users] set global variable > >Jan Fousek wrote: >> Hi all, >> is there any possibility of setting the global variables from outside of asterisk? >> Like manager api or something like that. >> >> Thanks a lot >> >> not sure about current svn trunk, > >but in the past you could set a channel var with > >action: SetVar >channel: Zap/49-1 >variable: SOMEVAR=SOMESTRING > >but required a channel > >action: GetVar will get channel or global vars > >you should be able to easily mod manager.c to use ast_get_channel_by_name >code if channel is passed, and use pbx_builtin_setglobalvar otherwise. > >sad thing is setvar is depeciated in favor of set. (reminder, no clue what current svn trunk has) > >i hope this helps > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set global variable
Hi all, is there any possibility of setting the global variables from outside of asterisk? Like manager api or something like that. Thanks a lot ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and spandsp
Hi, I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates the connection, recieves some info about the sender and dies. Result is not valid tif file of size 330 or 334 B. Debug messages are shown only sometimes and it ends with something like FLOW Fast carrier training failed FLOW Fast carrier down. Libtiff 3.7.1, asterisk 1.2.9.1, spandsp 0.0.2pre21 installed manually on gentoo. Has anyone made asterisk to recieve faxes this way? (or any other, I'm slowly running out of ideas) I'd be grateful for any reply Jan Fousek ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and spandsp
Hi, I'm trying to get my asterisk recieve faxes via rxfax and spandsp. It creates the connection, recieves some info about the sender and dies. Result is not valid tif file of size 330 or 334 B. Debug messages are shown only sometimes and it ends with something like FLOW Fast carrier training failed FLOW Fast carrier down. Libtiff 3.7.1, asterisk 1.2.9.1, spandsp 0.0.2pre21 installed manually on gentoo. Has anyone made asterisk to recieve faxes this way? (or any other, I'm slowly running out of ideas) I'd be grateful for any reply Jan Fousek ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: SIP Debug to file - Is it possible?
tcpdump -i eth0 -s 1500 -w ./dump ‘udp port 5060′ creates dump of communication on port 5060 (default sip) __ > Od: [EMAIL PROTECTED] > Komu: "Asterisk Users Mailing List - Non-Commercial > Discussion", "Asterisk Users Mailing List - > Non-Commercial Discussion" > Datum: 22.08.2006 05:18 > Předmět: RE: [asterisk-users] Re: SIP Debug to file - Is it possible? > >ngrep is also good if you only want to see SIP traffic and filter all the lower level stuff. > > -Original Message- > From: Brandon Galbraith [mailto:[EMAIL PROTECTED] > Sent: Mon 8/21/2006 8:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: > Subject: Re: [asterisk-users] Re: SIP Debug to file - Is it possible? > > > Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, not so sure about IAX though. > > -brandon > > > On 8/21/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote: > > Christopher Aloi wrote: > > Hello List - > > > > > > I'm a big fan of call traces to diagnose a problem; I > often use > > "pri set debug file X" to write PRI traces out to a file, > anyone > > know of a similar method of saving IP traces (SIP,IAX) to > a file? > > > > Anyone have any ngrep scripts that do the trick? > > > tcpdump :) ? > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Brandon Galbraith > Email: [EMAIL PROTECTED] > AIM: brandong00 > Voice: 630.400.6992 > "A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jitterbuffer on SIP
Hi, same by me, the patch affects app_rxfax, app_txfax and the G.726 codec from the spandsp. However, it doesn't link it to the libspandsp properly, asterisk complains: "undefined symbol: g726_encode". I added to modules.conf the line noload => codec_g726.so and asterisk comes up again. Thanks for a link to patch. Jan Fousek __ > Od: [EMAIL PROTECTED] > Komu: asterisk-users@lists.digium.com > Datum: 09.08.2006 23:26 > Předmět: [asterisk-users] Jitterbuffer on SIP > >Thank You Patrick, > >After some minor problems in some file paths I had success compiling. > >The only problem was the codec_g726 witch does an illegal call and Asterisk >doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the >modules directory and asterisk came up. > >I´m going to test it now. > >Thanks again!! > > > > >On Tue, 2006-08-08 at 10:45 -0300, Thierry Querette wrote: >> > Hi, >> > >> > Is that a way to patch a running asterisk 1.2.9.1 instalation with the >> > experimental SIP Jitterbuffer support ? >> >> Yes, see http://www.asterisk-backports.org >> >> http://asterisk-backports.org/downloads/ast_jb-1.2.9.1+rtp-keep-jb+fax >> +g726.patch >> >> The jb seems to work fine on my setup (with low usage). >> >> Regards, >> Patrick >> > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk dosenot compile
Hi, what about upgrading qmake? see http://lists.digium.com/pipermail/asterisk-dev/2006-July/021599.html Just guessing... __ > Od: [EMAIL PROTECTED] > Komu: asterisk-users@lists.digium.com > Datum: 04.08.2006 10:03 > Předmět: [asterisk-users] asterisk dosenot compile > >Hello friends, >I am trying to install asterisk. I downloaded the latest development branch from digium thru svn. I get an error in the make which says:- > [LD] codec_gsm.o gsm/lib/libgsm.a -> codec_gsm.so > [CC] codec_ilbc.c -> codec_ilbc.o >make[2]: Entering directory `/home/install/asterisk/codecs/ilbc' >make[2]: *** virtual memory exhausted. Stop. >make[2]: Leaving directory `/home/install/asterisk/codecs/ilbc' >make[1]: *** [ilbc/libilbc.a] Error 2 >make[1]: Leaving directory `/home/install/asterisk/codecs' >make: *** [codecs] Error 2 > >It seems the codec ilibc has a problem. I tried touch the file ilbc.o and ilbc.so. But it wouldnot help. Please suggest how do I go further with this? >Thankyou all in advance. > > > > > >With warm regards. > >Vivek J. Joshi. > >[EMAIL PROTECTED] >Trikon electronics Pvt. Ltd. > >All science is either physics or stamp collecting. >-- Ernest Rutherford > > > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ateus Easy gate call progress
Hi all, has anybody any experience with Ateus Easy Gate connected via Digium card to asterisk? It works fine for me except it doesn't pass the caller id and the hangup detection is quite slow. Are there some tips how to shorten the hangup delay? Thanks. Jan Fousek ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk fails to register, when the full logging is turned on
Hi all, my asterisk is registering for two accounts, each by different provider. Whenever I turn the full logging on, * fails to register for one of the accounts while the second one remains registered. Is this a known issue to somebody, or should I bugreport? Have a nice day Jan Fousek ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: Dropping incompatible voice frame
Hi, I saw the same error today. It appeared when the fax got transfered from s to fax context. Haven't seen sollution yet (turn of the detection, if the line is fax only and you are sure, nobody would like to make normal call to it. Hope somebody finds a fix Jan Fousek __ > Od: [EMAIL PROTECTED] > Komu: > Datum: 18.07.2006 20:42 > Předmět: [asterisk-users] Error: Dropping incompatible voice frame > >Hello, > >I get this error message when trying to route an incoming fax from a packet based T1 to an EICON board that is connected to an external fax & voice mail server. >Voice calls route to this external server with no error. Both fax and voice calls that come in a channelized T1 also route to this external server with no errors. >I am on 1.2.7.1 > >Jul 13 13:19:56 NOTICE[24867]: channel.c:1904 ast_read: Dropping incompatible voice frame on CAPI/PRI1/XX-14a of format slin since our native format has changed to ulaw > >I did find a reference to something similar in Mantis issue tracker number 0004101. >I am not technical enough to know if this is the same issue. Any help to explain what is the problem or how to fix it is greatly appreciated. >Thank you, >Tim > > > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax Got hangup
Hi all,
I'm trying to setup asterisk and spandsp to recieve fax transmissions. I
got Asterisk to detect fax calls, it even tries to communicate, but the
other side doesn't seem to send the main data. Instead it ends the
communication with hangup. Have anybody got an idea?
Thanks a lot.
Jan Fousek
This is a relevant part of the log:
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing
Answer("SIP/420543254384-b5dd", "") in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing
AbsoluteTimeout("SIP/420543254384-b5dd", "35") in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Set Absolute Timeout to 35
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing
Set("SIP/420543254384-b5dd",
"FAXFILE=/var/spool/asterisk-fax/1153245510.15.tif") in new stack
Jul 18 19:58:30 VERBOSE[20671] logger.c: -- Executing
RxFAX("SIP/420543254384-b5dd",
"/var/spool/asterisk-fax/1153245510.15.tif|debug") in new stack
Jul 18 19:58:30 DEBUG[31032] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Response 23395782: Match Found
Jul 18 19:58:32 DEBUG[31032] chan_sip.c: Auto destroying call '[EMAIL
PROTECTED]'
Jul 18 19:58:33 DEBUG[31032] chan_sip.c: Auto destroying call '[EMAIL
PROTECTED]'
Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 1 to 4
Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: FLOW >>> DIS:Jul 18 19:58:33
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: 00Jul
18 19:58:33 DEBUG[20671] app_rxfax.c: ceJul 18 19:58:33 DEBUG[20671]
app_rxfax.c: f4Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:33
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: 81Jul
18 19:58:33 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:33 DEBUG[20671]
app_rxfax.c: 80Jul 18 19:58:33 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:33
DEBUG[20671] app_rxfax.c: 18Jul 18 19:58:33 DEBUG[20671] app_rxfax.c:
Jul 18 19:58:35 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:35 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:36 DEBUG[20671] app_rxfax.c: FLOW HDLC carrier up
Jul 18 19:58:36 DEBUG[20671] app_rxfax.c: FLOW HDLC framing OK
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW <<< ???:Jul 18 19:58:37
DEBUG[20671] app_rxfax.c: 1aJul 18 19:58:37 DEBUG[20671] app_rxfax.c:
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW ??? with final frame tag
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW In state 9
Jul 18 19:58:37 DEBUG[20671] app_rxfax.c: FLOW HDLC carrier down
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: FLOW >>> DIS:Jul 18 19:58:38
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: 00Jul
18 19:58:38 DEBUG[20671] app_rxfax.c: ceJul 18 19:58:38 DEBUG[20671]
app_rxfax.c: f4Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:38
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: 81Jul
18 19:58:38 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:38 DEBUG[20671]
app_rxfax.c: 80Jul 18 19:58:38 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:38
DEBUG[20671] app_rxfax.c: 18Jul 18 19:58:38 DEBUG[20671] app_rxfax.c:
Jul 18 19:58:40 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:40 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: FLOW >>> DIS:Jul 18 19:58:43
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: 00Jul
18 19:58:43 DEBUG[20671] app_rxfax.c: ceJul 18 19:58:43 DEBUG[20671]
app_rxfax.c: f4Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:43
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: 81Jul
18 19:58:43 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:43 DEBUG[20671]
app_rxfax.c: 80Jul 18 19:58:43 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:43
DEBUG[20671] app_rxfax.c: 18Jul 18 19:58:43 DEBUG[20671] app_rxfax.c:
Jul 18 19:58:45 DEBUG[20671] app_rxfax.c: FLOW HDLC underflow in state 9
Jul 18 19:58:45 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 4 to 3
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW T4 timeout in state 9
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW Changed from phase 3 to 4
Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: FLOW >>> DIS:Jul 18 19:58:48
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: 00Jul
18 19:58:48 DEBUG[20671] app_rxfax.c: ceJul 18 19:58:48 DEBUG[20671]
app_rxfax.c: f4Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:48
DEBUG[20671] app_rxfax.c: 80Jul 18 19:58:48 DEBUG[20671] app_rxfax.c: 81Jul
18 19:58
