RE: [Asterisk-Users] Asterisk Receptionist manager program.
Hi, -Original Message- If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. Aye! I'd love to have a look. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing multiple extensions
On Tue, May 25, 2004 at 05:53:55PM -0500, Roger spake thusly: Thanks for the reply - I have version cell phone service. I did a work around and called my cell phone via IAX2 as opposed to the zaptel channels. This works and all 3 extensions ring w/ no problem. I am having the exact same problem. I am trying to simultaneously ring my SIP phone on my desk and my cell via a Zap interface connected to my PRI. The phone dialed with the Zap interface rings and the SIP phone never rings even though on the console it says both are being dialed. I considered what the previous poster said about the cell phone company picking up before ringing my cell phone and as a test tried calling my home phone on a POTS line. That phone always rang and the SIP didn't. So it seems that the call made through the Zap interface will always go through and nothing else will. I have considered turning callprogress=yes in my zapata.conf file but the explanation of that option in the example file makes me hesitant to use it. Plus it mentioned using it only with FXS analog lines. I'm on a PRI so I have real digital signalling. If you were able to use an IAX channel to some other provider who is surely just connected to another PRI and it worked just fine then it sounds like perhaps this is a bug or at least a mis-feature because the intermediate Asterisk box is then shielding you from the apparent answer on the PRI without passing the answer signal back to you until the phone is really answered. How there could be two such apparent answers is beyond me though. -- Tracy Reed The attachment is a digital signature. http://copilotconsulting.com More info: http://copilotconsulting.com/sig pgprzhbq8P8xN.pgp Description: PGP signature
Re: [Asterisk-Users] Disable blind xfer
--On Friday, May 28, 2004 2:57 pm -0400 Timothy R. McKee [EMAIL PROTECTED] wrote: My SIP users need to transmit the # key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. This is roughly the same issue as the double hash transfer I implemented for analogue phones connecting through an ATA. Search for that. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Example: Caller*ID Fixup Macro for use with DIDs
Here is a macro I wrote to deal with sites that have a mixture of
extensions with DID numbers and extensions without DID numbers. This
macro is USA/Canada specific but could be adapted for the dial plans in
other countries. This macro uses features that are not in CVS -stable.
You must use CVS -head for this macro to work.
[default]
exten = _91NXXNXX,1,Macro(callerid-fixup,5045551212)
exten = _91NXXNXX,2,Dial(Zap/g1/${EXTEN:1})
exten = 2101,1,Macro(callerid-fixup)
exten = 2101,2,Dial(SIP/2101)
[macro-callerid-fixup]
;
; This macro assumes the following:
; Extensions without a DID number have a CallerID number that is 4 digits
; Extensions with a DID number have their CallerID number set to their 10 digit DID
; The last 4 digits of a DID number is the same as the extension number
;
; This macro does the following if it is called without any parameters:
; If the Caller*ID number is 4 digits, do nothing
; If the Caller*ID number is not 4 digits, strip off the first 6 digits
;
; This macro does the following if it is called with a parameter
; If the Caller*ID number is 4 digits, reset the Caller*ID number to be the value of
the parameter
; If the Caller*ID number is not 4 digits, do nothing
;
exten = s,1,NoOp(Entering Macro callerid-fixup)
exten = s,2,NoOp(Variable CALLERIDNUM is equal to ${CALLERIDNUM})
exten = s,3,GoToIf($[${ARG1} != ]?external,1)
exten = s,4,GoToIf($[${ARG1} = ]?internal,1)
exten = internal,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
exten = internal,2,SetCIDNum(${CALLERIDNUM:6})
exten = internal,3,GoTo(f,1)
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
exten = external,2,SetCIDNum(${ARG1},a)
exten = external,3,GoTo(f,1)
exten = f,1,NoOp(Variable CALLERIDNUM is equal to ${CALLERIDNUM})
exten = f,2,NoOp(Leaving Macro callerid-fixup)
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Asterisk Receptionist manager program.
Dear, Kyle Hagan wrote: We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. Thats a good experience for me too, please do the needful. make it fast, keep me updated [EMAIL PROTECTED] We are considering writing a SIP client build into the program at a later time. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Example: Caller*ID Fixup Macro for use with DIDs
Fixed an error in the macro The line:
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
should be changed to
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} != 4]?f,1)
--Eric
On Sat, 2004-05-29 at 02:49, Eric Wieling wrote:
Here is a macro I wrote to deal with sites that have a mixture of
extensions with DID numbers and extensions without DID numbers. This
macro is USA/Canada specific but could be adapted for the dial plans in
other countries. This macro uses features that are not in CVS -stable.
You must use CVS -head for this macro to work.
[default]
exten = _91NXXNXX,1,Macro(callerid-fixup,5045551212)
exten = _91NXXNXX,2,Dial(Zap/g1/${EXTEN:1})
exten = 2101,1,Macro(callerid-fixup)
exten = 2101,2,Dial(SIP/2101)
[macro-callerid-fixup]
;
; This macro assumes the following:
; Extensions without a DID number have a CallerID number that is 4 digits
; Extensions with a DID number have their CallerID number set to their 10 digit
DID
; The last 4 digits of a DID number is the same as the extension number
;
; This macro does the following if it is called without any parameters:
; If the Caller*ID number is 4 digits, do nothing
; If the Caller*ID number is not 4 digits, strip off the first 6 digits
;
; This macro does the following if it is called with a parameter
; If the Caller*ID number is 4 digits, reset the Caller*ID number to be the value
of the parameter
; If the Caller*ID number is not 4 digits, do nothing
;
exten = s,1,NoOp(Entering Macro callerid-fixup)
exten = s,2,NoOp(Variable CALLERIDNUM is equal to ${CALLERIDNUM})
exten = s,3,GoToIf($[${ARG1} != ]?external,1)
exten = s,4,GoToIf($[${ARG1} = ]?internal,1)
exten = internal,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
exten = internal,2,SetCIDNum(${CALLERIDNUM:6})
exten = internal,3,GoTo(f,1)
exten = external,1,GoToIf($[${LEN(${CALLERIDNUM})} = 4]?f,1)
exten = external,2,SetCIDNum(${ARG1},a)
exten = external,3,GoTo(f,1)
exten = f,1,NoOp(Variable CALLERIDNUM is equal to ${CALLERIDNUM})
exten = f,2,NoOp(Leaving Macro callerid-fixup)
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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[Asterisk-Users] zaptel startup issues solved
I was having trouble getting zaptel to startup. Thought I'd share my experience since I saw others with the same issue and no solutions. I'm running Fedora FC2. Symptoms were: xtcfg -vv that says that it's unable to open master device /etc/dev/ctl modprobe zaptel was also failing /var/log/messages had messages like: Apr 12 18:48:45 asterisk kernel: zaptel: version magic '2.6.5-1.315custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.315 686 REGPARM 4KSTACKS gcc-3.3' The solution was to go into /usr/src/linux-2.6/include/linux/version.h and change the UTS_RELEASE string from '2.6.5-1.315custom' to '2.6.5-1.315' Then rebuild zaptel, install and run again. Obviously I chose the custom option when I setup the OS. Is there someway to fix the zaptel code to not be so picky? -mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ringing sound on GS phones
Use r option in your Dial command. - Original Message - From: joachim [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 2:45 AM Subject: Re: [Asterisk-Users] No ringing sound on GS phones Make sure to use CVS-head and you'll get ringing. At 23:51 28/05/2004, you wrote: On Fri, 2004-05-28 at 16:54, Stefan de Konink wrote: The same problems occurs at our Red Hat system after the upgrade from 0.7.2 to 0.9.0. I didn't tryed the Grandstream phones, but our SIP enabled Cisco 79xx's. I doubt its the grandstream phones. We have a testbed here like this: GS102 -- Asterisk 0.7.2 -- (IAX2) -- Asterisk 0.9.0 -- GS102 Calling from 0.9.0 - 0.7.2 we get ringback. Calling the other way we get no ringback. Before upgrading to 0.9.0 we got ringback in both directions. -- Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaptel startup issues solved
In article [EMAIL PROTECTED], Mike Stupak [EMAIL PROTECTED] wrote: I was having trouble getting zaptel to startup. Thought I'd share my experience since I saw others with the same issue and no solutions. I'm running Fedora FC2. Symptoms were: xtcfg -vv that says that it's unable to open master device /etc/dev/ctl modprobe zaptel was also failing /var/log/messages had messages like: Apr 12 18:48:45 asterisk kernel: zaptel: version magic '2.6.5-1.315custom 686 REGPARM 4KSTACKS gcc-3.3' should be '2.6.5-1.315 686 REGPARM 4KSTACKS gcc-3.3' The solution was to go into /usr/src/linux-2.6/include/linux/version.h and change the UTS_RELEASE string from '2.6.5-1.315custom' to '2.6.5-1.315' Then rebuild zaptel, install and run again. Obviously I chose the custom option when I setup the OS. Is there someway to fix the zaptel code to not be so picky? I think it expects the kernel source tree to match the running kernel. If you had built a new kernel called 2.6.5-1.315custom and then booted from it, you would probably have built zaptel successfully. I think :-) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp wont compile.
then run ldconfig or restart your machine...:) W - Original Message - From: Sam Bingner [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 12:26 AM Subject: RE: [Asterisk-Users] spandsp wont compile. Add the path to it to /etc/ld.so.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Friday, May 28, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Downgrading Asterisk
It seems the choppy (and almost unusable) audio in Head is only impacting some cisco users, and since these problems are not impacting the few that can read code, use cisco phones, and are impacted, we're stuck with the problem. The problem seems to be very evasive, however switching the iax2 links to use only iLBC (and not gsm) has corrected issues for some. In my case I never had any problems doing SIP to SIP - even from non-cisco to the cisco phones or visa versa. I only had problems with calls between the cisco and the POTS world through chan_capi. This problem has been solved by the patch to rtp.c posted in this mailing list about a month ago (an if statement commented out). With that patch HEAD works perfectly ok for me (ahem - I _hate_ writing that). Lars, By commenting out that if statement, you are disabling the function that ties the timestamps together, and will have to keep doing that every time you update code. That's a problem bypass and not a real fix. (Also, if you're not very carefull with that, cvs update will likely fail to update the rtp.c code.) I've asked several times why it's important to tie the timestamps together, and as of today no one has hinted or even guessed at the reason for it. Therefore, pure guess is that Mark has some architectual reason (probably related to datastream distortion) for it that will bite us later on. Since I'm not a programmer and can't read the code worth a damn, I'd have to venture a guess that iax2-rtp has had many issues addressed, but other channels such as capi-rtp still need to be cleaned up. Since the noise level seems to be oriented around iax2 and capi, its probably fair to assume other channels are working as expected. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel startup issues solved
Hi. cut Obviously I chose the custom option when I setup the OS. Is there someway to fix the zaptel code to not be so picky? I think it expects the kernel source tree to match the running kernel. If you had built a new kernel called 2.6.5-1.315custom and then booted from it, you would probably have built zaptel successfully. I think :-) sure, that happens because kernel-source package from fedora has the kernel version set to blahcustom, as long as many other rh versions. the solutions are 2: * use a plain, vanilla kernel * as Tony suggests, rebuild the kernel first, install it and then build zaptel Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel startup issues solved
On Sat, 2004-05-29 at 14:15 +0200, Brancaleoni Matteo wrote: sure, that happens because kernel-source package from fedora has the kernel version set to blahcustom, as long as many other rh versions. Mandrake does this as well. Great idea if you want to increase the noise on a list. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
CVS HEAD from about 1 week ago. TDM30P and call through Nufone. I was talking and wanted to park the call and move to another phone to pick it up. I hit #701 instead of #700 though -- after a pause, I got a fast busy and the call was gone. When I called the person back, she said that Alison told HER that 701 was an invalid extension. I should have heard that though, not her. If I dial #700 *I* hear Alison, like I should... For this reason I don't think it's a dialplan issue, but has anyone else seen this? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
Hi, this is no bug. When you want to park a call just hit #700. Alison will then tell you on which extension the call was parked. To pick up this call just dial the announced extension (e.g. 701). When you press #700 while in a call you connect this call to the call parked at this extension and, if no call is parked, he will talk to alison. But he would hear that there is no call parked and not that it is a invalid extension. I hope I got it rigth, if not I'm sorry. Bye - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] I hit #701 instead of #700 though -- after a pause, I got a fast busy and the call was gone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp wont compile.
I got it to load BUT now i get when i try to load the module. localhost*CLI load app_rxfax.so localhost*CLI May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize Unable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize On Fri, 2004-05-28 at 22:04, Mark Musone wrote: Make sure that /usr/local/lib is in your /etc/ld.so.conf After you do a make install of spandsp. Also make sure you run ldconfig to update the librarys -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp wont compile.
/etc/ld.so.conf /usr/X11R6/lib /usr/lib/qt3/lib /usr/local/libUnable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize /usr/local/lib/libtiff /usr/lib/asterisk/modules the mods compiled BUT now won't load. On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote: add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
On Saturday 29 May 2004 09:28, FastJack wrote: this is no bug. When you want to park a call just hit #700. Alison will then tell you on which extension the call was parked. To pick up this call just dial the announced extension (e.g. 701). When you press #700 while in a call you connect this call to the call parked at this extension and, if no call is parked, he will talk to alison. But he would hear that there is no call parked and not that it is a invalid extension. I hope I got it rigth, if not I'm sorry. I think you misunderstood. I have the asterisk system. *I* should be the one who hears Alison at all times, not the remote party. To reiterate. Me (on Zap/3-1 calling Them via Nufone): #701 [ I hear nothing for a few seconds, then fast busy ] [ REMOTE PARTY hears Alison say 701 is an invalid extension ] I should he hearing Alison say that, and get reconnected to the call (or at the very least back to * so I can enter another extension), not disconnected. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
Its a known bug I posted a patch eons back but i can't find now with that fscking pathetic search on mantis wish http://www.google.com/custom?sitesearch=bugs.digium.com could work ok Found it http://bugs.digium.com/bug_view_page.php?bug_id=487 - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 7:06 AM Subject: Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me) On Saturday 29 May 2004 09:28, FastJack wrote: this is no bug. When you want to park a call just hit #700. Alison will then tell you on which extension the call was parked. To pick up this call just dial the announced extension (e.g. 701). When you press #700 while in a call you connect this call to the call parked at this extension and, if no call is parked, he will talk to alison. But he would hear that there is no call parked and not that it is a invalid extension. I hope I got it rigth, if not I'm sorry. I think you misunderstood. I have the asterisk system. *I* should be the one who hears Alison at all times, not the remote party. To reiterate. Me (on Zap/3-1 calling Them via Nufone): #701 [ I hear nothing for a few seconds, then fast busy ] [ REMOTE PARTY hears Alison say 701 is an invalid extension ] I should he hearing Alison say that, and get reconnected to the call (or at the very least back to * so I can enter another extension), not disconnected. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me)
Or you can use ValetParking :P bkw - Original Message - From: TC [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 8:33 AM Subject: Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me) Its a known bug I posted a patch eons back but i can't find now with that fscking pathetic search on mantis wish http://www.google.com/custom?sitesearch=bugs.digium.com could work ok Found it http://bugs.digium.com/bug_view_page.php?bug_id=487 - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 7:06 AM Subject: Re: [Asterisk-Users] transfer bug (#701 - remote party hears alison, not me) On Saturday 29 May 2004 09:28, FastJack wrote: this is no bug. When you want to park a call just hit #700. Alison will then tell you on which extension the call was parked. To pick up this call just dial the announced extension (e.g. 701). When you press #700 while in a call you connect this call to the call parked at this extension and, if no call is parked, he will talk to alison. But he would hear that there is no call parked and not that it is a invalid extension. I hope I got it rigth, if not I'm sorry. I think you misunderstood. I have the asterisk system. *I* should be the one who hears Alison at all times, not the remote party. To reiterate. Me (on Zap/3-1 calling Them via Nufone): #701 [ I hear nothing for a few seconds, then fast busy ] [ REMOTE PARTY hears Alison say 701 is an invalid extension ] I should he hearing Alison say that, and get reconnected to the call (or at the very least back to * so I can enter another extension), not disconnected. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay when routing PSTN - IAXy dect phone
Just setup *, got a developers kit FXO where the incoming/outgoing pstn is plugged in. I've then got an IAXy that is plugged into a Philips DECT phone. * is setup so that the [bell] section rings the phone - exten = s,1,Dial(IAX2/myuser,30) What's happening when someone calls my number is that the phone rings 3 times before you hear the dect phone ringing. Anybody got any ideas as its driving me crazy? Logs from * (don't worry about the agi script here I've tried it with those lines commented and get the same results): May 29 15:32:10 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2 (Ring/Answered)... May 29 15:32:12 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2 (Ring/Answered)... May 29 15:32:13 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2 (Ring/Answered)... May 29 15:32:15 NOTICE[-1373635664]: chan_zap.c:4797 ss_thread: Got event 2 (Ring/Answered)... -- Executing AGI(Zap/1-1, mclid.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/mclid.agi -- AGI Script Executing Application: (SETCIDNAME) Options: (0143211) -- AGI Script Executing Application: (SETCIDNUM) Options: (0143211) -- AGI Script mclid.agi completed, returning 0 -- Executing Dial(Zap/1-1, IAX2/myuser|30) in new stack -- Called myuser -- Call accepted by 192.168.0.60 (format ULAW) -- Format for call is ULAW -- IAX2[myuser]/4 is ringing -- IAX2[myuser]/4 answered Zap/1-1 -- Hungup 'IAX2[myuser]/4' Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Galaxy Voice
Has anyone successfully used Galaxy Voice with Asterisk? I am getting the following SIP errors repeated whether it is or isn't behind NAT. May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 111 (Critical Request) May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again asterisk2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Galaxy Voice
First off they are not ERRORS they are NOTICE and WARNING. bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 10:26 AM Subject: [Asterisk-Users] Galaxy Voice Has anyone successfully used Galaxy Voice with Asterisk? I am getting the following SIP errors repeated whether it is or isn't behind NAT. May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 111 (Critical Request) May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again asterisk2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Galaxy Voice
Also I think someone posted a galaxy voice config example on the mailing list a few weeks back.. have you searched google yet? bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 11:04 AM Subject: RE: [Asterisk-Users] Galaxy Voice I deeply apologize for the incorrect statement, thanks for taking the time to point out the error...your help is appreciated. -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Saturday, May 29, 2004 1:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Galaxy Voice First off they are not ERRORS they are NOTICE and WARNING. bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 10:26 AM Subject: [Asterisk-Users] Galaxy Voice Has anyone successfully used Galaxy Voice with Asterisk? I am getting the following SIP errors repeated whether it is or isn't behind NAT. May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 111 (Critical Request) May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again asterisk2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PlayTones problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: -- Executing Answer([EMAIL PROTECTED]/4, ) in new stack -- Executing Playtones([EMAIL PROTECTED]/4, Busy) in new stack -- Executing Hangup([EMAIL PROTECTED]/4, ) in new stack == Spawn extension (icepage, 123, 3) exited non-zero on '[EMAIL PROTECTED]/4' May 29 20:00:10 NOTICE[21526]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM -- Hungup '[EMAIL PROTECTED]/4' I don't have any FXS-hardware so I can't try it with a real phone. I am running the (stable) asterisk version from cvs from 3 days ago. Sincerely, Bartek - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuNDJWYjaxM2wIe4RAtS1AKCwMqmaKILwzLg9ZnKx0+uDEw5drwCdFqqv vUBt3kLL7jVDsnWVrKYGr9w= =P8PB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Caller ID with BT CD50
In article [EMAIL PROTECTED], Tony Hoyle [EMAIL PROTECTED] wrote: Kevin Walsh wrote: I downloaded the latest version of your patch, from your website, and it works perfectly. I had waited until I had some time available because I thought I'd have to play around with it for a while. Great. Just need to make sure that it still works for US lines and it's all set. There's some debate whether to use this patch or to wait for one that uses line reversal/guard tone detection... there is the slight problem that the X100P can't detect line reversal so it'd mean everyone upgrading their hardware... still, I have what 'works for me' and will continue hosting it for a while whatever happens. Is that the X100P generically, including the X101P? I would include both algorithms - the line reversal one for the hardware that can do it, and your current one for those that can't. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Beep Sound
Hi! Does anyone have a more clear beep tone for the voicemail? Try Playtones(): http://www.voip-info.org/wiki-Asterisk+cmd+Playtones Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd behaviour with asterisk -rx
Hello, I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold 1incoming 2incoming 4incoming 5incoming debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s Members: SIP/agent2 (dynamic) has taken 11 calls (last was 386 secs ago) SIP/agent1 (dynamic) has taken 16 calls (last was 1566 secs ago) Callers: 1. Zap/4-1 (wait: 0:08) debian:/# I'm on asterisk 1.0_stable - has this been fixed in head, is it a known issue? I was unable to find anything about it in a search of previous list posts... regards, Julien Levi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Caller ID with BT CD50
Tony Mountifield wrote: Is that the X100P generically, including the X101P? I think so (same driver)... they're just software modems really, so there's no need for them to have that kind of detection on-chip. I would include both algorithms - the line reversal one for the hardware that can do it, and your current one for those that can't. Me too - the current patch could also be used to do DTMF caller ID without too much work (there isn't a line reversal in the specs for that, you just have to look for valid digits). I'll probably do some tidying up (change ukcallerid to callerid=uk as it's neater).. the zaptel side though is stable. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PlayTones problem
On Sat, 29 May 2004, Bartek Kania wrote: I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. My extensions.conf always include a wait statement after playtones. Try it: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Wait(2) ; play busy tone for 2 seconds exten = 123,4,Hangup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp wont compile.
Your most likely compiling against one tiff library version, but loading up another... Do a: ldd app_rxfax.so to see what tiff library it's compiled against, and then also try to find all the places where libtiff is on your machine and remove the incorrect one.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Saturday, May 29, 2004 6:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp wont compile. /etc/ld.so.conf /usr/X11R6/lib /usr/lib/qt3/lib /usr/local/libUnable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize /usr/local/lib/libtiff /usr/lib/asterisk/modules the mods compiled BUT now won't load. On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote: add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Hi I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: why don't you use the manager interface? it's much better... Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PlayTones problem
yep use exten = 123,1,Answer exten = 123,2,Busy bkw - Original Message - From: Bartek Kania [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 12:04 PM Subject: [Asterisk-Users] PlayTones problem -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: -- Executing Answer([EMAIL PROTECTED]/4, ) in new stack -- Executing Playtones([EMAIL PROTECTED]/4, Busy) in new stack -- Executing Hangup([EMAIL PROTECTED]/4, ) in new stack == Spawn extension (icepage, 123, 3) exited non-zero on '[EMAIL PROTECTED]/4' May 29 20:00:10 NOTICE[21526]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM -- Hungup '[EMAIL PROTECTED]/4' I don't have any FXS-hardware so I can't try it with a real phone. I am running the (stable) asterisk version from cvs from 3 days ago. Sincerely, Bartek - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuNDJWYjaxM2wIe4RAtS1AKCwMqmaKILwzLg9ZnKx0+uDEw5drwCdFqqv vUBt3kLL7jVDsnWVrKYGr9w= =P8PB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Galaxy Voice
Yes, I did a search and have what I think is the correct configuration. I did a google search and I didn't see much. I was successful in getting it to work both inbound and outbound with the exception of the notices and warnings. The config I am using is: [galaxyvoice] nat=yes port=5060 fromuser=12345678 fromdomain=216.229.127.40 username=12345678 type=friend secret=12345678 auth=md5 host=216.229.127.40 ;defaultip=216.229.127.40 reinvite=no canreinvite=no dtmfmode=rfc2833 context=inbound-galaxy qualify=yes disallow=all allow=gsm allow=ulaw callerid=12345678 incominglimit=2 defaultexpirey=60 -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Saturday, May 29, 2004 2:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Galaxy Voice Also I think someone posted a galaxy voice config example on the mailing list a few weeks back.. have you searched google yet? bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 11:04 AM Subject: RE: [Asterisk-Users] Galaxy Voice I deeply apologize for the incorrect statement, thanks for taking the time to point out the error...your help is appreciated. -Original Message- From: brian k. west [mailto:[EMAIL PROTECTED] Sent: Saturday, May 29, 2004 1:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Galaxy Voice First off they are not ERRORS they are NOTICE and WARNING. bkw - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 10:26 AM Subject: [Asterisk-Users] Galaxy Voice Has anyone successfully used Galaxy Voice with Asterisk? I am getting the following SIP errors repeated whether it is or isn't behind NAT. May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 111 (Critical Request) May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again asterisk2*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Known issue bkw - Original Message - From: Julien Levi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 12:12 PM Subject: [Asterisk-Users] Odd behaviour with asterisk -rx Hello, I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold 1incoming 2incoming 4incoming 5incoming debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s Members: SIP/agent2 (dynamic) has taken 11 calls (last was 386 secs ago) SIP/agent1 (dynamic) has taken 16 calls (last was 1566 secs ago) Callers: 1. Zap/4-1 (wait: 0:08) debian:/# I'm on asterisk 1.0_stable - has this been fixed in head, is it a known issue? I was unable to find anything about it in a search of previous list posts... regards, Julien Levi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Yes, it's a known issue on the bug tracker (#1110), but no solution has been found to date, afaik. Julien Levi wrote: Hello, I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold 1incoming 2incoming 4incoming 5incoming debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s Members: SIP/agent2 (dynamic) has taken 11 calls (last was 386 secs ago) SIP/agent1 (dynamic) has taken 16 calls (last was 1566 secs ago) Callers: 1. Zap/4-1 (wait: 0:08) debian:/# I'm on asterisk 1.0_stable - has this been fixed in head, is it a known issue? I was unable to find anything about it in a search of previous list posts... regards, Julien Levi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Caller ID with BT CD50
Tony Hoyle wrote: I'll probably do some tidying up (change ukcallerid to callerid=uk as it's neater).. the zaptel side though is stable. OK... now uses usecallerid=uk (or usecallerid=us for symmetry). Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
its not really a critical issue... wonder when someone will take the time and fix it. :P bkw - Original Message - From: Karl Brose [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 12:29 PM Subject: Re: [Asterisk-Users] Odd behaviour with asterisk -rx Yes, it's a known issue on the bug tracker (#1110), but no solution has been found to date, afaik. Julien Levi wrote: Hello, I was planning to use the output of asterisk -rx show queues in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold debian:/# asterisk -rx zap show channels Chan Extension Context Language MusicOnHold 1incoming 2incoming 4incoming 5incoming debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s debian:/# asterisk -rx show queues bookingsqhas 1 calls (max 100) in 'ringall' strategy (9s holdtime), C:28, A:3, SL:57.1% within 20s Members: SIP/agent2 (dynamic) has taken 11 calls (last was 386 secs ago) SIP/agent1 (dynamic) has taken 16 calls (last was 1566 secs ago) Callers: 1. Zap/4-1 (wait: 0:08) debian:/# I'm on asterisk 1.0_stable - has this been fixed in head, is it a known issue? I was unable to find anything about it in a search of previous list posts... regards, Julien Levi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PlayTones problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sat, 29 May 2004, Hermann Wecke wrote: On Sat, 29 May 2004, Bartek Kania wrote: I am having problems with the PlayTones application and VoIP softphones. But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. My extensions.conf always include a wait statement after playtones. Try it: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Wait(2); play busy tone for 2 seconds exten = 123,4,Hangup Thanks a bunch! I feel a little stupid for missing something like this though =) But I still get the following message on the console: May 29 21:01:58 NOTICE[22550]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM Can I just ignore it? /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuN6hWYjaxM2wIe4RAjqKAKCM37PIM+7Bb1OdOJIzOsWfocA0nACgncga dPgH1FX+jHVW8S67fM9jpVU= =tFbW -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP extension
Hi,
my VoIP provider routes my main phone number and all extensions to the
same sip account.
In the sip header of the invite message is the To: field that shows me
which extension the caller dialed ...
for extension 0:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
[...]
To: sip:[EMAIL PROTECTED]
for extension 10:
To: sip:[EMAIL PROTECTED]
How can I use this extension information in my dialplan?
I have tried ${RDNIS}
(http://lists.digium.com/pipermail/asterisk-users/2004-March/
041264.html) but this doesn´t work (RDNIS is empty).
I use CVS-HEAD-05/22/04
TIA,
Mike
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Re: [Asterisk-Users] bug or feature?
At 7:33 AM -0700 on 5/26/04, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? [catching up on 800 -user posts - sorry for delay] Nobody on the list suggested this method that I saw: Use the Background application, but play silence. You'll notice in the asterisk-sounds directory (the additional package) there is a directory called silence which contains 10 files ranging from 1 to 10 seconds of silence. Works the same as Wait from the user's perspective (they hear nothing) but lets the user type keys. That's why I made those files; it's only a slightly ugly hack, and it works quite well. :-) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bug or feature?
John Todd wrote: At 7:33 AM -0700 on 5/26/04, Maveric wrote: I've noticed that when i pass a wait in an exten = that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? [catching up on 800 -user posts - sorry for delay] Nobody on the list suggested this method that I saw: Use the Background application, but play silence. You'll notice in the asterisk-sounds directory (the additional package) there is a directory called silence which contains 10 files ranging from 1 to 10 seconds of silence. Works the same as Wait from the user's perspective (they hear nothing) but lets the user type keys. That's why I made those files; it's only a slightly ugly hack, and it works quite well. :-) Only slighly ? :-) -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura stun settings
AJ Grinnell wrote: Well, it worked for 1 call, but now I am back to getting half a ring from the ATA and then nothing. I am only seeing one rtp packet recieved per call. Any other ideas? Does it work ok if canreinvite=no ?? To be honest I have not experimented much with canreinvite. All our Sipura subs register with SER and not Asterisk. STUN works fine for us with the settings I sent you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andres Sent: Wednesday, May 26, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura stun settings AJ Grinnell wrote: I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk server and STUN server are outside the firewall on a public network. I would like the Sipuras to be able to reinvite, so I set canreinvite=yes in my sip.conf, and set the STUN server under the SIP tab in the Sipuras. However, I am not able to hear the other caller (the Sipura is not recieving RTP packets, it is sending just fine). Am I missing something on the Sipura config? I am not sure what all of the VIA options mean, and which ones I should use. Cant find any good info out there, can someone hrer help me out? Thank you. You need these settings: Substitute_VIA_Addr Yes ; STUN_Enable Yes ; NAT_Mapping_Enable[1] Yes ; NAT_Keep_Alive_Enable[1] Yes ; STUN_Test_Enable Yes; and of course define your STUN Server. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned by Arialink for dangerous content and is believed to be clean. For more information please email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net Providing Wholesale Florida SIP/IAX2 Termination for US$0.01/minute ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream ringtone maker (was Re: Grandstream v1.0.4.68 firmware)
Hi! Excellent!! Philipp See near the bottom for the interesting bit :-) OK, while composing this post I decided to write a perl program to read a uLaw stream on standard input and create a suitable header, writing the result to an output file. It can be found at http://www.softins.co.uk/makering.pl.txt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 29/05/2004 at 13:52 brian k. west wrote: its not really a critical issue... wonder when someone will take the time and fix it. :P bkw to you bkw_ .. it's actually quite important to some of us... a bit like DTMF callerid :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Caller ID with BT CD50
On 29/05/2004 at 19:16 Tony Hoyle wrote: Me too - the current patch could also be used to do DTMF caller ID without too much work (there isn't a line reversal in the specs for that, you just have to look for valid digits). I'll probably do some tidying up (change ukcallerid to callerid=uk as it's neater).. the zaptel side though is stable. Tony Tony, there's a bounty (although it's not much but it's better than a poke in the eye with a sharp stick) for DTMF callerid (some of us have been bitching about it for ages)... http://bugs.digium.com/bug_view_page.php?bug_id=0001265 For reference Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Caller ID with BT CD50
Andy Powell wrote: there's a bounty (although it's not much but it's better than a poke in the eye with a sharp stick) for DTMF callerid (some of us have been bitching about it for ages)... http://bugs.digium.com/bug_view_page.php?bug_id=0001265 I can't do it unfortunately, as I'm not in a country with DTMF caller ID, but it should be as simple(!) as: 1. Add 'usecallerid=europe' or something suitable, set use_callerid == 3 2. In the code that tests caller id, just if use_callerid == 3 instead of passing the history buffer through the callerid state machine, pass it through the dtmf decoder (ast_dsp_new() and friends). It's harder than doing it for the UK but not so much harder that someone couldn't knock it together in a day or two... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
Accually you can issue the cli commands via manager and get full outputs! (Most people dont know that) bkw - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 2:59 PM Subject: Re: [Asterisk-Users] Odd behaviour with asterisk -rx On 29/05/2004 at 13:52 brian k. west wrote: its not really a critical issue... wonder when someone will take the time and fix it. :P bkw to you bkw_ .. it's actually quite important to some of us... a bit like DTMF callerid :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd behaviour with asterisk -rx
On 29/05/2004 at 16:49 brian k. west wrote: Accually you can issue the cli commands via manager and get full outputs! (Most people dont know that) bkw yes you can, but you have to have blocking=yes ... and I'm still waiting for info on what the implications of doing this are.. eg if the manager session is disconnected mid transmission... etc no one appears to know... or care ... or both Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs problem with TDM04B ?
Hi, On 28-May-04, at 6:16 AM, Rich Adamson wrote: I there a problem with CVS ? My card TDM04B does not want to answer calls on 2 ports. Strange. Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from about 5/24) and it will work again. Mark is aware of the problem. Do you have a bug# we can track? I'm about to deploy this card in a production env - I was going to pull the latest drivers, but if bug is still active, I'll stick with pre 5/25. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extracting country code from a number
Hi Does anyone know of an algorythm to extract the country code from a number. I understand that the country codes are of different length and there is no fixed length of local area code or phone numbers. I am sure there is a way, if not how to telephone switches handle them Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling under Suse 9.0
Hello Everybody... probably this is an FAQ item but I can't find it anywhere so here it goes. I am trying to compile the latest CVS release of Asterisk under Suse 9.0 zaptel and libpri compile without any problems but when I do a 'make install' in asterisk, it compiles ok until it reaches the cdr directory. There I get a number of error messages mainly whenever CONNECTION_BAD and CONNECTION_OK appear. Does anybody know what I am doing wrong? Or if there is something else I need to compile/include before trying the asterisk directory? Many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling under Suse 9.0
On Sat, 2004-05-29 at 18:01, Vassilis Konstantinou wrote:
I am trying to compile the latest CVS release of Asterisk under Suse 9.0
zaptel and libpri compile without any problems but when I do a 'make
install' in asterisk, it compiles ok until it reaches
the cdr directory.
There I get a number of error messages mainly whenever CONNECTION_BAD and
CONNECTION_OK appear.
Does anybody know what I am doing wrong? Or if there is something else I
need to compile/include before trying the
asterisk directory?
Looks like Asterisk, for some reason, thinks you want to use the
Postgress features for CDR and Voicemail. I don't know why. You don't
need these features to use Asterisk.
[EMAIL PROTECTED] asterisk]# grep -r CONNECTION_BAD /home/software/asterisk/asterisk
/home/software/asterisk/asterisk/apps/app_sql_postgres.c:if (PQstatus(karoto)
== CONNECTION_BAD) {
/home/software/asterisk/asterisk/apps/app_voicemail.c: if (PQstatus(dbhandler) ==
CONNECTION_BAD) {
/home/software/asterisk/asterisk/cdr/cdr_pgsql.c: if (PQstatus(conn) !=
CONNECTION_BAD) {
/home/software/asterisk/asterisk/cdr/cdr_pgsql.c: if (PQstatus(conn) !=
CONNECTION_BAD) {
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] cvs problem with TDM04B ?
Yes there is a problem. Pull an older copy of wcfxs.c in zaptel (from about 5/24) and it will work again. Mark is aware of the problem. Do you have a bug# we can track? I'm about to deploy this card in a production env - I was going to pull the latest drivers, but if bug is still active, I'll stick with pre 5/25. There was a problem, no bug report created, Mark was made aware of it, and either fixed it or removed the broken code (don't know which). It's working now. The wcfxs.c file is now dated May 26 13:22 (53081 bytes) from Head. From what I can tell, the driver for the tdm/fxo card has additional work to be done, but it is working now. (ie, expect more changes.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iConnectHere broken?
Title: iConnectHere broken? I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a Unsupported Media error even though I'm still using ulaw and alaw. It stopped working with my softphone, my Grandstreams, my Snom, etc. Searched the lists for a recent discussion of this but couldn't find it. Any help? Thx!
[Asterisk-Users] Asterisk - Zaptel - DIGIUM x 4 T1
Hi all, Anybody try to configure Asterisk and Digium card on linux-2.6.5-1.358 FEDORA CORE2 ? Making the zaptel getting error: storage size. Thanks, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Webmin Module in download directory
Is the webmin module in the download directory of asterisk on site still maintained. I can't get it to install with the latest webmin - 1.140, it says it can't find the module.info file in the webmin module. Thanks, Nicholas Ruddick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iConnectHere broken?
I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a Unsupported Media error even though I'm still using ulaw and alaw. It stopped working with my softphone, my Grandstreams, my Snom, etc. Searched the lists for a recent discussion of this but couldn't find it. I'm not a user of that service, but I'd suggest using Head instead of Stable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Webmin Module in download directory
is there a webmin add in and where I would like to test it... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extracting country code from a number
search google... rgagon posted something to -dev that does just this a few months back. bkw - Original Message - From: usedcanon [EMAIL PROTECTED] To: Asterisk users [EMAIL PROTECTED] Sent: Saturday, May 29, 2004 5:01 PM Subject: [Asterisk-Users] extracting country code from a number Hi Does anyone know of an algorythm to extract the country code from a number. I understand that the country codes are of different length and there is no fixed length of local area code or phone numbers. I am sure there is a way, if not how to telephone switches handle them Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E164.org Updates
Firstly we've setup a SIP proxy that uses e164.org to do enum lookups, also rather then issuing people with yet more numbers they have to remember we've coded up a watered down version of e164.org for people that would just like to have a single SIP phone rather then run their own PABX. http://www.Like2Fone.Com for more details on that service. The plan is to get people hooked on VoIP so they will setup Asterisk at a later date and make the most out of an enum lookup service. We've also started to import blocks of numbers into e164.org and allow people to manage the blocks themselves via the web interface. Currently we're not issuing NS records at this stage as it limits smaller and single number blocks and we end up with the situation the ITU has where carriers don't want to proceed so they don't sub-allocate DNS. Long term we're thinking of doing this in a secondary zone and allowing people to order which preference they prefer or only allocating NS records on small blocks of number ranges. So if you have a large chunk of number space and are happy to advertise it over the net in DNS, email me the range and the associated VoIP URL off list. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
I would love to test it, but I need some help with the zapata config settings. In the US you can have up to 4 numbers on a line each with a ring pattern. The ring patterns are: Long ring - 1st number (this is a normal ring - like if you didn't have distinctive ringing at all) Double ring - 2nd number Short short long - 3rd number Short long short - 4th number I have 3 phone numbers on one line so can test all but the last. How would I configure in zapata.conf? From the example I would think that dring1 = the double ring and dring2 = short short long and if it didn't match either of those it would be the normal long ring? Or is that referring to two different things and dring1 = long, dring2 = double and dring3 = ssl and dring4 = sls (assuming there is a dring3 and dring4)? ;dring1=95,0,0 ;dring1context=internal1 ;dring2=325,95,0 ;dring2context=internal2 ; If no pattern is matched here is where we go. ;context=default ;channel = 1 Then I am a little confused about the values after the = can you tell me what they would be for the above ring patterns? I see others that have tested (in Europe?) using: dring1 = 367,0,0 dring1context = incoming-pstn-personal dring2 = 247,0,0 dring2context = incoming-pstn-business I apologize if this is documented somewhere, but I couldn't find it in any searches in the wiki. Ben On Friday, May 28, 2004, at 06:30 PM, Tony Hoyle wrote: Tony Hoyle wrote: I've changed the patch to fix the buffer overrun, plus a hack to only look for the dring values you specify, thus: btw. It would be great if someone in the US can test these changes to make sure I haven't broken the CID/DR on their side by doing this. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Receptionist manager program.
I run 2 small call centers, would love to help Zac -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Friday, May 28, 2004 11:33 AM To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist manager program. We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. We are considering writing a SIP client build into the program at a later time. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom and multiple lines
How do I get the lights to work correctly on a SNOM 200 when I configure it for more than one line? The lights stay on solid, although the buttons work correctly for making calls. Thanks in advance. Dennis Engdahl SnowCrest, Inc. www.snowcrest.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
