[asterisk-users] DAHDI, and 64 bit machine
Hi All; To download, compile and install DAHDI, do I need to download the both (dahdi-kernel and dahdi-tools) If yes, then do I need to do the compilation and installation command for each package? What is the method to download, compile and install the both packages as one package? By the way: Why there is dahdi-kernel and dahdi-tools? In other words, for what the kernel is used and for what the tools is used? And why they called the dahdi-kernel in that name (related it to the kernel? to which kernel?) About the 64 bit machine: If my machine is 64 bit, does that effect on selecting the DAHDI and the Asterisk version? Or All work for 64 and 32? I know that in codecs, there is a difference in the machine is 64 or 32, but what in the Asterisk and DAHDI? regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO clock
Dear sir I build a daughter card to interface the FXO module with blackfin537 stamp board, but unfortunately I can`t get Dial tone or any other signalling via fxo port.however the fxo module has been detected on the board. my daughter card work with clock= 2.048MHZ, and i configure asterisk with USA signalling.I suspect that the frequency of 2.048MHZ (form E1 TDM frame ) is not suitable for USA signalling as USA work with standard T1 TDM . am I right or not,are that could make a problem. Thanks for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI, and 64 bit machine
On Sat, Jun 6, 2009 at 5:33 AM, bilal ghayyad bilmar...@yahoo.com wrote: To download, compile and install DAHDI, do I need to download the both (dahdi-kernel and dahdi-tools) If yes, then do I need to do the compilation and installation command for each package? What is the method to download, compile and install the both packages as one package? depends on your OS. You may be able to find compatible pre-compiled packages for your distribution. But compiling will work too. When you download the packages there are instructions inside in conveniently named files like README. By the way: Why there is dahdi-kernel and dahdi-tools? In other words, for what the kernel is used and for what the tools is used? And why they called the dahdi-kernel in that name (related it to the kernel? to which kernel?) the kernel package contains the kernel drivers source code. the tools package contains the tools source code. Drivers make the hardware talk to the kernel. Tools make the drivers tell the hardware what you want them to do, query status, etc. About the 64 bit machine: If my machine is 64 bit, does that effect on selecting the DAHDI and the Asterisk version? Or All work for 64 and 32? I know that in codecs, there is a difference in the machine is 64 or 32, but what in the Asterisk and DAHDI? No. The compilation process auto-detects your system settings and builds correctly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What does it mean rc in the release version
Hi All; When I find the rc in the release name dahdi-linux-2.2.0-rc5.tar.gz, then what does it mean the rc5? Which is better, to select dahdi-linux-2.2.0-rc5.tar.gz or to select dahdi-linux-2.1.0.tar.gz? I am afraid that rc means still not finally finished and has bugs? Any advise? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO clock
On Sat, 6 Jun 2009, Ayman Hendawy wrote: Dear sir I build a daughter card to interface the FXO module with blackfin537 stamp board, but unfortunately I can`t get Dial tone or any other signalling via fxo port.however the fxo module has been detected on the board. my daughter card work with clock= 2.048MHZ, and i configure asterisk with USA signalling.I suspect that the frequency of 2.048MHZ (form E1 TDM frame ) is not suitable for USA signalling as USA work with standard T1 TDM . am I right or not,are that could make a problem. Thanks for your help. You need to integrate the FXS module if you want dial tone from it. Are you trying to build a digital interface or analog? j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What does it mean rc in the release version
When I find the rc in the release name dahdi-linux-2.2.0-rc5.tar.gz, then what does it mean the rc5? Release Candidate. Which is better, to select dahdi-linux-2.2.0-rc5.tar.gz or to select dahdi-linux-2.1.0.tar.gz? I am afraid that rc means still not finally finished and has bugs? Any advise? RC's aren't officially released. You may have a higher chance of encountering bugs. If you are concerned with stability to stick a released version. jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP/AGI/SetVar Issue
Steve Edwards schrieb:
If I follow the protocol:
printf(STREAM FILE \hit\ \\\n);
fflush(stdout);
fgets(response, sizeof(response), stdin);
printf(STREAM FILE \hit\ \\\n);
fflush(stdout);
fgets(response, sizeof(response), stdin);
it works everywhere, every time.
PHP:
# enable implicit flushing of the output buffer.
# required for CentOS/RHEL because on that platform the php
# binary is php-cgi (which is wrong) instead of php-cli so
# fFlush(STDOUT) is not available
# (http://bugs.centos.org/view.php?id=1633)
ini_set('implicit_flush', 1);
ob_implicit_flush(1);
echo 'STREAM FILE hit' ,\n;
//fFlush(STDOUT);
while (fgetc(STDIN) !== false) {} // read (and ignore) STDIN
echo 'STREAM FILE hit' ,\n;
//fFlush(STDOUT);
while (fgetc(STDIN) !== false) {} // read (and ignore) STDIN
As I and others have suggested, stand on the shoulders of others -- please
use an established library. You will save time and hair. And your code
will be easier to write and maintain:
agi_stream_file(hit, );
agi_stream_file(hit, );
True, if you're able to find a library which properly escapes
strings etc. for the language of your choice.
Philipp Kempgen
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Re: [asterisk-users] Digium Fax Driver
If you're building RPMS, it's just a matter of e.g. mock -r fedora-10-i386 asterisk-1.6.1.0-0.1.fc10.src.rpm. I think mock works with dpkg-based systems too. It's incredibly handy. You can use it to build by hand as well (without a package manager), but I haven't tried that. As an alternative, you can use any virtualization/compartmentalization software. OpenVZ, Linux-vserver, KVM, Xen, VMWare... I just think mock is easier. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI, and 64 bit machine
bilal ghayyad schrieb: To download, compile and install DAHDI, do I need to download the both (dahdi-kernel and dahdi-tools) Yes. You can either get http://downloads.asterisk.org/pub/telephony/dahdi-linux/ and http://downloads.asterisk.org/pub/telephony/dahdi-tools/ or use the combined tarball http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ If yes, then do I need to do the compilation and installation command for each package? Yes if you download both tarballs separately. The complete tarball automates it for you. What is the method to download, compile and install the both packages as one package? cd /usr/src wget dahdi-...tar.gz tar -xvzf dahdi-...tar.gz cd dahdi-...tar.gz less README* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we have put in a temp OpenVOX tdm400 card and it works perfectly. As soon as we swap that and use Sangoma via wanrouter we get crosstalk. For example, if an existing call is happening and a new internal to external call or vise versa happens, they can hear each other, even just to IVR. Any ideas? All wiring has been checked and this *does not*, I repeat, *does not* happen with the Sangoma card. So what ever explaination we come up with, that fact remains and we get stumped. Oh, the card and four fxo modules have been completely replaced and 64bit has been compiled in the wanrouter driver and Sangoma tech support have ran out of suggestions. We have also tried going down to 2gb on the 64bit system too. Hopefully 32bit will work, but we have other clients on 64bit with Sangoma and they work. What is the Sangoma latest stable 64bit driver doing! Thanks, Gavin. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we have put in a temp OpenVOX tdm400 card and it works perfectly. As soon as we swap that and use Sangoma via wanrouter we get crosstalk. For example, if an existing call is happening and a new internal to external call or vise versa happens, they can hear each other, even just to IVR. Any ideas? All wiring has been checked and this *does not*, I repeat, *does not* happen with the Sangoma card. So what ever explaination we come up with, that fact remains and we get stumped. Oh, the card and four fxo modules have been completely replaced and 64bit has been compiled in the wanrouter driver and Sangoma tech support have ran out of suggestions. We have also tried going down to 2gb on the 64bit system too. Hopefully 32bit will work, but we have other clients on 64bit with Sangoma and they work. What is the Sangoma latest stable 64bit driver doing! Thanks, Gavin. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Teliax: where's the space in CALLERID(num) from?
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
[DialOut] ; subroutine for dialing out.
exten = s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} )
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,GotoIf($[${CALLERID(num)} 200]?dial-out)
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
exten = s,n(dial-out),Dial(${First-Preferred-Out}/${ARG1})
...
Here's the CLI when dialing out:
-- Executing [917yyyx...@longdistance:2] Gosub(SIP/178-081c52a0,
DialOut,s,1(917yyy)) in new stack
-- Executing [...@dialout:1] NoOp(SIP/178-081c52a0, Context:
DialOut called with outgoing number 917yyy )
in new stack
-- Executing [...@dialout:2] NoOp(SIP/178-081c52a0, 178) in
new stack
-- Executing [...@dialout:3] GotoIf(SIP/178-081c52a0, 0?dial-out)
in new stack
-- Executing [...@dialout:4] Set(SIP/178-081c52a0,
CALLERID(num)=xxx178 ) in new stack
-- Executing [...@dialout:5] Dial(SIP/178-081c52a0,
IAX2/zz...@nyc.teliax.net/917yyy) in new stack
Now I think the reason that teliax isn't seeing my callerid is that it's
looking for a valid 10 digit number. But when * sets the callerid in
s...@dialout:4 there's a trailing space(and yes, in real life I use an
actual 10 digit number), so teliax is probably getting 11 characters as
the callerid.
But CALLERID(num) does not start with a trailing space - see
s...@dialout:2, so concatenating it with the rest of the number shouldn't
create a space. Right?
sean
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Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Currently we have put in a temp OpenVOX tdm400 card and it works perfectly. As soon as we swap that and use Sangoma via wanrouter we get crosstalk. For example, if an existing call is happening and a new internal to external call or vise versa happens, they can hear each other, even just to IVR. How often does this happen? (the cross-talk) every single call? is easy to reproduce? Any ideas? All wiring has been checked and this *does not*, I repeat, *does not* happen with the Sangoma card. So what ever explaination we come up with, that fact remains and we get stumped. You meant that this does not happen with the OpenVox card, didn't you? otherwise, you lost me. If you can easily reproduce this, I'd be interested in look into it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI, and 64 bit machine
Hi David; But why in Dahdi there is a kernel package and tool package while in zaptel there were not such thing, we were download zaptel only. Why they separated them in Dahdi? Now, regarding to compilation and installation: 1) Why when using the complete package, I do not need to do the ./configure while I need to do this if I am trying to compile and install the dahdi-tools? 2) Why I need to write make all when I am trying to compile and install the dahdi-linux-complete? In other words, why to use make all and does not use make only? Regards Bilal - To download, compile and install DAHDI, do I need to download the both (dahdi-kernel and dahdi-tools) If yes, then do I need to do the compilation and installation command for each package? What is the method to download, compile and install the both packages as one package? depends on your OS. You may be able to find compatible pre-compiled packages for your distribution. But compiling will work too. When you download the packages there are instructions inside in conveniently named files like README. By the way: Why there is dahdi-kernel and dahdi-tools? In other words, for what the kernel is used and for what the tools is used? And why they called the dahdi-kernel in that name (related it to the kernel? to which kernel?) the kernel package contains the kernel drivers source code. the tools package contains the tools source code. Drivers make the hardware talk to the kernel. Tools make the drivers tell the hardware what you want them to do, query status, etc. About the 64 bit machine: If my machine is 64 bit, does that effect on selecting the DAHDI and the Asterisk version? Or All work for 64 and 32? I know that in codecs, there is a difference in the machine is 64 or 32, but what in the Asterisk and DAHDI? No. The compilation process auto-detects your system settings and builds correctly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?
sean darcy schrieb:
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
-- Executing [...@dialout:4] Set(SIP/178-081c52a0,
CALLERID(num)=xxx178 ) in new stack
But when * sets the callerid in
s...@dialout:4 there's a trailing space
But CALLERID(num) does not start with a trailing space - see
s...@dialout:2, so concatenating it with the rest of the number shouldn't
create a space. Right?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] DAHDI, and 64 bit machine
bilal ghayyad schrieb: Hi David; But why in Dahdi there is a kernel package and tool package while in zaptel there were not such thing, we were download zaptel only. Both -kernel and -tools were in Zaptel as well. They were just not separated in 2 tarballs. Why they separated them in Dahdi? Cleanliness? 1) Why when using the complete package, I do not need to do the ./configure while I need to do this if I am trying to compile and install the dahdi-tools? Because the Makefile in the -complete tarball automatically calls ./configure for you. 2) Why I need to write make all when I am trying to compile and install the dahdi-linux-complete? In other words, why to use make all and does not use make only? Maybe only make works. Give it a try. If make is called without a target it tries to run the first target in the Makefile which usually happens to be all. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI, and 64 bit machine
Philipp Kempgen schrieb: bilal ghayyad schrieb: 2) Why I need to write make all when I am trying to compile and install the dahdi-linux-complete? In other words, why to use make all and does not use make only? Maybe only make works. Give it a try. If make doesn't work, where's the problem with make all? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI, and 64 bit machine
On Sat, Jun 06, 2009 at 02:33:28AM -0700, bilal ghayyad wrote: By the way: Why there is dahdi-kernel and dahdi-tools? In other words, for what the kernel is used and for what the tools is used? And why they called the dahdi-kernel in that name (related it to the kernel? to which kernel?) The basic idea is that dahdi-linux is the linux-specific part whereas dahdi-tools should be os-independent. I'm not really sure that this is the case with the dahdi-perl scripts (that rely heavily on the procfs and sysfs interfaces) and the ppp module in dahdi tools. About the 64 bit machine: If my machine is 64 bit, does that effect on selecting the DAHDI and the Asterisk version? Or All work for 64 and 32? I know that in codecs, there is a difference in the machine is 64 or 32, but what in the Asterisk and DAHDI? As I regularily use DAHDI drivers on my 64bit laptop, I can say it works rather well :-) One limitation: if dahdi is built as a module for a 64bit kernel, it will not support 32bit userspace clients. This is normally not something that you do, though. https://issues.asterisk.org/view.php?id=14808 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Every call as soon as the sangoma card is live. Speak to Konrad on your techdesk for more info. Thanks. On 06/06/2009, Moises Silva moises.si...@gmail.com wrote: Currently we have put in a temp OpenVOX tdm400 card and it works perfectly. As soon as we swap that and use Sangoma via wanrouter we get crosstalk. For example, if an existing call is happening and a new internal to external call or vise versa happens, they can hear each other, even just to IVR. How often does this happen? (the cross-talk) every single call? is easy to reproduce? Any ideas? All wiring has been checked and this *does not*, I repeat, *does not* happen with the Sangoma card. So what ever explaination we come up with, that fact remains and we get stumped. You meant that this does not happen with the OpenVox card, didn't you? otherwise, you lost me. If you can easily reproduce this, I'd be interested in look into it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How run AsyncAGI commands in background
Hi, Asterisk 1.4.18 AsyncAGI patch from //http://moythreads.com/testasync2.diff http://moythreads.com/testasync2.diff// Regards Jose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How run AsyncAGI commands in background
then it should work, create a *simple* extensions.conf and pastebin it along with instructions so I can try to reproduce. On Sat, Jun 6, 2009 at 5:02 PM, Jose Ariascyr2...@gmail.com wrote: Hi, Asterisk 1.4.18 AsyncAGI patch from http://moythreads.com/testasync2.diff Regards Jose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
On Sat, Jun 6, 2009 at 3:18 PM, Gavin Henrygavin.he...@gmail.com wrote: Every call as soon as the sangoma card is live. Speak to Konrad on your techdesk for more info. Thanks. I'll speak with him on Monday. However if you can provide more information before Monday I will be able to think beforehand on this matter. So please confirm this. If you get an incoming call and send it to Playback(demo-congrats) and then receive a second call and send it to Playback(tt-monkeys), both callers will listen both demo-congrats and tt-monkeys sounds? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How run AsyncAGI commands in background
Jose Arias schrieb: Hi, Asterisk 1.4.18 AsyncAGI patch from //http://moythreads.com/testasync2.diff http://moythreads.com/testasync2.diff// Regards So what? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How run AsyncAGI commands in background
On Sat, Jun 6, 2009 at 7:18 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Jose Arias schrieb: Hi, Asterisk 1.4.18 AsyncAGI patch from //http://moythreads.com/testasync2.diff http://moythreads.com/testasync2.diff// Regards So what? What do you mean with so what?, if you have not been involved in the conversation you would not understand. http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer call from analog telephone
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Daniel Bareiro wrote:
As I've commented in a previous message, after dial *60 (of *600 to Echo
test), I obtain like a tone cut in three parts followed of a continuous tone,
causing that I'm incapable to dial the extension completely. The
waitfordigit appears after to hangup. The cell_number seems to be some
number that I has dial previously. Testing again with a SIP extension, this
problem does not happen.
Also it draws attention to me that the DTMF has a duration of 0ms.
It is peculiar... after to have a restart of Asterisk, I can dial without
problems to *600. This is Asterisk log corresponding to the successful
communication with the extension:
- --
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:30] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:03:30] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:03:31] DTMF[28905]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:03:31] DTMF[28905]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Executing [*...@phones:1] Answer(DAHDI/2-1, ) in new stack
[Jun 4 23:03:31] DEBUG[28905]: chan_dahdi.c:3174 dahdi_answer: Took
DAHDI/2-1 off hook
-- Executing [*...@phones:2] Playback(DAHDI/2-1, demo-echotest) in
new stack
-- DAHDI/2-1Playing 'demo-echotest' (language 'es')
== Spawn extension (phones, *600, 2) exited non-zero on 'DAHDI/2-1'
-- Hungup 'DAHDI/2-1'
- --
As you will see, the duration is always of 0 ms (also when I dial to the cell
phone). After this I make several tests. To dial from cell phone to the analog
phone and I did not have problems in to call immediately to *600 after to have
dial to the cell phone in each opportunity. But if from my extension 201 I
dial the analog phone and after that from my analog phone I dial to *600, it
happens the same of problem of not to be able to dial beyond *60. Log of the
CLI for this situation is the following one:
- --
[Jun 4 23:08:45] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '*'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:45] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '*' on DAHDI/2-1
[Jun 4 23:08:45] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '*' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '6'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '6' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2229 __ast_read: DTMF end '0'
received on DAHDI/2-1, duration 0 ms
[Jun 4 23:08:46] DTMF[29017]: channel.c:2271 __ast_read: DTMF end accepted
without begin '0' on DAHDI/2-1
[Jun 4 23:08:46] DTMF[29017]: channel.c:2282 __ast_read: DTMF end
passthrough '0' on DAHDI/2-1
-- Blacklisting number 201
[Jun 4 23:08:54] DEBUG[29017]: chan_dahdi.c:6244 ss_thread: waitfordigit
returned 0...
-- Hungup 'DAHDI/2-1'
- --
Testing some more I could verify than if I changed the number for echo test to
*700 instead of *600, the problem of not being able to dial beyond *60
disappears. Investigating a little in Internet and reading the source code, I
found the following in the line 2834 of chan_mgcp.c file:
- -
2834 } else if (!ast_strlen_zero(p-lastcallerid) !strcmp(p-dtmf_buf,
*60)) {
2835 if (option_verbose 2) {
2836 ast_verbose(VERBOSE_PREFIX_3 Blacklisting number
%s\n, p-lastcallerid);
2837 }
2838 res =
