[asterisk-users] Asterisk news :: Next release of Asterisk will be 1.8 Long Term Support
Dear Asterisk community, Yesterday, Russell Bryant finally made up his mind and confirmed on the asterisk-dev mailing list that the next release of Asterisk will be 1.8, which will also be a Long Term Support (LTS) release. This also means that the 1.4 is now officially classed as a LTS release too. I feel that this is a very good solution for the whole Asterisk community and that we all will benefit from it. I have personally not been happy with the 1.6.x release schedule, which has been very misunderstood and has confused a large group of users. Hopefully, we can now continue with a release schedule that the world understands and that makes sense for everyone. While I understand the need for releasing quicker than we've done in the past, the detail about the naming, the actual release numbers (1.6.0, 1.6.1 etc) was very hard to explain to people. With years of experience of doing Asterisk and VoIP training, I have a lot of respect of the need of being able to easily explain things, from configuration details to release schedule... Now we, the Asterisk community, need to focus quickly on the new release and plan what's going in there. If you have code for new features lying around (as I have tons of in various branches of my svn repository), now is the proper time to step forward, contribute it to the bug tracker and get it evaluated, discussed and maybe finally included in Asterisk. Whatever goes into 1.8 at release time, will be what we will have for production use for a long time. We also ask you to dedicate time during next year to help the Asterisk project with testing. You don't have to be a developer to test - and we need tests of everything from documentation to configuration and technichal issues. We don't have all of the equipment you have, we don't have your dialplans, we don't have all the applications you integrate Asterisk with. If Asterisk is important to your organization, please make sure that you dedicate time during the first half of 2010 to do regular testing of the new release betas and release candidates. We do need your help to make Asterisk 1.8 a good release, worthy to replace the 1.4 as a new LTS release. If you're a member of a Linux or Asterisk group, please help in organizing Asterisk 1.8 test-partys. If you need help with ideas, please contact our community liason, John Todd. Meeting other Asterisk users, testing stuff together is one of the best ways to expand your knowledge of Asterisk. Sharing ideas and how-to's in real time while setting up test labs and scenarious is really, really fun. Here in Sweden, where I live, we have half a meter of snow and very cold weather. The days are very short and I've tried to brighten up the darkness by decorating my house with a large amount of blinking lamps. No, they're not SIP compliant using Subscribe/notify, sorry. That may be a project for a test - to see how many phones with subscriptions one Asterisk can carry. If that works out well, maybe my house's blinking lamps can be powered by SIP and Asterisk 1.8 next year :-) This is the final workday for me before Christmas. Tomorrow, I'll bake the traditional Swedish Christmas ham in the owen and then continue with the Christmas bread. I just love Christmas - the food, the family and friends, the gifts and, well, the food again :-) I wish you all a Merry Christmas and a Happy New Year! /Olle ___ -- 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] Every one Busy Problem
On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote: Hello When ever i try to use Dial DAHDI / SIP i get the following warning and nothing happens and Asterisk moves to next instruction Even i know that channel is free no one else is using it [Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) What was the Dial line you used? What type of DAHDI device? FXS? FXO? PRI? Wasn't the number simply busy? -- 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] anonymous calls code
Hi C F,
I tried but does not work. It seems that my telco (telecom) does not
accept any number with a leading '*'.
Asterisk CLI returns busy:
empty_chan_in_stack: cannot empty channel 255
as if the channel were busy...but it works if I connect a normal phone
(and it worked with the old analog PBX).
I'm a bit puzzled...
Giorgio
C F wrote:
You would have to create a dialplan for it.
If your provider expects *67 (which is the case here with I/CLEC POTS)
then you would create something like:
exten = _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
In the case of PRI you would use:
exten = _*67[2-9]XX,1,SetCallerPres(prohib)
exten = _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
does anybody know how to make on-demand anonymous calls? I've tried code
*67# before the number to call but it is working with some providers only.
Any hints?
Thank you.
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Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.comwrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether we are using the GNU C compiler... (cached) yes
checking whether gcc accepts -g... (cached) yes
checking for gcc option to accept ISO C89... (cached) none needed
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl.exe... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking whether we are using the GNU C++ compiler... no
checking whether g++ accepts -g... no
checking how to run the C preprocessor... gcc -E
checking how to run the C++ preprocessor... /lib/cpp
configure: error: in `/usr/local/asterisk-1.4.26':
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
Do you have gcc and company installed? gxx, g++?
--
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
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[asterisk-users] asterisk x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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Re: [asterisk-users] asterisk x-lite
Try tcpdump to see where RTP go from asterisk.
Configure your x-lite
Use stun server ?
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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Re: [asterisk-users] asterisk x-lite
It is a nat problem
François BERGANZ
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?
Doug, It doesn't respond to the INVITE - the trace says No response to the INVITE?. If the phone doesn't even ring it's probably not getting anything, which points to a problem with the router it's behind. How is the router set up to deliver SIP and RTP to the phone? On Tue, Dec 22, 2009 at 5:33 AM, Doug d...@natel.net wrote: At 00:46 12/21/2009, Alex Balashov wrote: A packet capture would be needed to illuminate the source of the problem. Thanks, Alex for your suggestion. Here is a link for the packet capture: http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txthttp://www.A7H.com/%7Estuph/TCPdump-2009-Dec-21-2304.txt I just don't see where the extension responds to the INVITE. What would prevent that? By the way, I have a bunch of phones behind this same router that work just fine on our old v1.2 system. On 12/21/2009 01:39 AM, Doug wrote: I've turned on NAT everywhere I can think, but even though I hear ringing on the calling phone (different system) the called phone does not ring. Has anyone bumped into this lately? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk x-lite
Where is your definition of codecs ??
On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone.
The softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten =
_1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There
are not any firewall or iptables rules. I tried the nat=yes
parameter but no changes.
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Re: [asterisk-users] anonymous calls code
Hi C F,
I solved the problem!! It was under my nose...
If you are interested the solution is here:
http://www.misdn.org/index.php/FAQ_chan_mISDN
The right section is: key pad elements
Giorgio Incantalupo
C F wrote:
You would have to create a dialplan for it.
If your provider expects *67 (which is the case here with I/CLEC POTS)
then you would create something like:
exten = _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
In the case of PRI you would use:
exten = _*67[2-9]XX,1,SetCallerPres(prohib)
exten = _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
does anybody know how to make on-demand anonymous calls? I've tried code
*67# before the number to call but it is working with some providers only.
Any hints?
Thank you.
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Re: [asterisk-users] Every one Busy Problem
Thanks Tzaffir, Acutally i reinstalled DAHDI Asterisk and every thing seem to work fine now. i am using TDM800P with 8 FXO ports. the Number wasnt busy and asterisk server can recieve calls through that channel but cant use that channel to dial out. As the problem is solved :) so what left to explain Cheers On Tue, Dec 22, 2009 at 2:21 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote: Hello When ever i try to use Dial DAHDI / SIP i get the following warning and nothing happens and Asterisk moves to next instruction Even i know that channel is free no one else is using it [Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) What was the Dial line you used? What type of DAHDI device? FXS? FXO? PRI? Wasn't the number simply busy? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- Best Regards Shakeel Abbas ___ -- 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] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
wrote:
And what is the output of the ./configure? Does it generate any
errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com
wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.com
wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether we are using the GNU C compiler... (cached) yes
checking whether gcc accepts -g... (cached) yes
checking for gcc option to accept ISO C89... (cached) none needed
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl.exe... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking whether we are using the GNU C++ compiler... no
checking whether g++ accepts -g... no
checking how to run the C preprocessor... gcc -E
checking how to run the C++ preprocessor... /lib/cpp
configure: error: in `/usr/local/asterisk-1.4.26':
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
Do you have gcc and company installed? gxx, g++?
--
Tzafrir Cohen
icq#16849755
jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Yes , I have g++ installed .
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[asterisk-users] Making a data connection with Asterisk
Hi all, We need to start obtaining some billing files from BT via a dial-up ISDN connection and I'm wondering if Asterisk is capable of doing this? I need to make an ISDN dial-up CHAP connection and, once connected, grab some files over FTP. Currently, our Asterisk box is connected to an ISDN30 with a Zaptel card. This may be a downright stupid question but I'm trying to find out if Asterisk can be of any use.. Any info or pointers in the right direction would be appreciated. Thanks, Will ___ -- 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] Available Agent on Queue
Hi,
I have a problem..I have 3 agents active in my queue..How can I get list
off all my active agents who was available when a call come?
in my extension.conf :
(1)exten = 6501,n,Queue(${EXTEN},ntT,,,1)
(2)exten = 6501,n,Queue(${EXTEN},ntT,,,25)
after the first one running, if i set like that in my extension.conf, in
queue_log I can see ringing agents before entering the second queue
1261475067|1261475061.106|6501|IRMA|RINGNOANSWER|1000
1261475067|1261475061.106|6501|UMIE|RINGNOANSWER|1000
1261475067|1261475061.106|6501|EGA|RINGNOANSWER|1000
Is there any simple dialplan or is there any available option to get
without exit the queue? I'm using Asterisk 1.4.26.2.
Best regards,
Daniel
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Re: [asterisk-users] Making a data connection with Asterisk
Have a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS On Tue, 22 Dec 2009 10:57:36 +, Will Payne w...@teambadger.co.uk wrote: Hi all, We need to start obtaining some billing files from BT via a dial-up ISDN connection and I'm wondering if Asterisk is capable of doing this? I need to make an ISDN dial-up CHAP connection and, once connected, grab some files over FTP. Currently, our Asterisk box is connected to an ISDN30 with a Zaptel card. This may be a downright stupid question but I'm trying to find out if Asterisk can be of any use.. Any info or pointers in the right direction would be appreciated. Thanks, Will ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
I recommend you follow the detailed install guide in this book and install all
the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
Thank you for contacting Kesher Communications Ltd.
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Company has taken reasonable precautions to ensure no viruses are present in
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made by a sender or by Kesher Communications Ltd. Recipient(s) takes
responsibility for any actions taken as a result of advice and recommendations
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: 22 December 2009 10:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen
tzafrir.co...@xorcom.commailto:tzafrir.co...@xorcom.com wrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby
wcse...@selbytech.commailto:wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi
motamed...@gmail.commailto:motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby
wcse...@selbytech.commailto:wcse...@selbytech.comwrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.commailto:motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.comhttp://www.selbytech.com/
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for gcc... (cached) gcc
checking whether
Re: [asterisk-users] Asterisk 1.2.14 - Play an audio or signal
Thanks a lot Alec, I´ll check
2009/12/22 Alec Davis siva...@paradise.net.nz
straight from our 1.6.1 dialplan, don't know about 1.2.14.
exten = s,n,Set(LIMIT_WARNING_FILE=beep)
exten = s,n,Set(LIMIT_TIMEOUT_FILE=call-terminated)
;terminate after 1 hour, start beep warnings at 10 minutes, every 5 minutes
exten =
s,n,Dial(${AVAILCHAN_NOSESSION}/${ARG2}#,,rL(360:300:30))
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Juan David Diaz
*Sent:* Tuesday, 22 December 2009 11:40 a.m.
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Asterisk 1.2.14 - Play an audio or signal
Good Day List Users,
Is there any way to play an audiofile or at least a beep into an
established call, I want to do this event each 3 minutes in the call, for
now I have a shell to get the call time and evaluate the 3 minutes.do
you know any way to play that sound?
I tried app_inject, it works really nice in asterisk 1.4.X releases; but
my PBX runs 1.2.14 and It can´t be upgraded (policy reasons).
Regards and Thanks every one.
--
Juan.
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--
Juan.
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[asterisk-users] AsteriskNow and language
Hi, I installed AsteriskNow and upgraded FreePBX to 2.6.0. In a sip extension definition, when I set language, it is not reported in the extensions_custom.conf file (eg language=xx). Am I missing something or is it not the right way to set language? BTW, is this a valid place for AsteriskNow questions? Dedicated mailing list seems dead. Thanks for answer -- Daniel ___ -- 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] Showing name of extension when calling
Hi! Is it possible, when placing a call that u see the name of the extension in your diplay? For example, 2 sip.conf entries: [971] callerid=Stefan [975] callerid=Magnus 975 calls 971 today 975 sees 971 in the display but would like to se: Stefan or just Stefan or... /Magnus ___ -- 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] E1 R2 Congestion Status
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk Digium E1 R2
ProtocolàCisco E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
configuration ?
=
Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on
rev-212-98-156-56 (pid = 3614)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [00223...@default:1] Dial(SIP/98.34.56.216-000e,
DAHDI/g1/00223344) in new stack
[Dec 22 06:02:49] WARNING[4756]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/98.34.56.216-000e' status is
'CONGESTION'
Cisco Gateway
show controller
Controller E1 slot(0)/port(0)
E1 Link is UP
No Alarm detected.
Applique type is Channelized E1.
Framing is CRC4, Line Code is HDB3.
Signalling type is R2-MFC.
0 Line Code Violations, 0 Framing Bit Errors
0 Far End Block Errors, 0 CRC Errors
signalling type = r2
clock source = slave
channel group 0 = 1-31
1 2 3
allocated timeslots = YYYNYYY
outgoing barred channel group =
channel order = ascending
b-channel negotiation = exclusive
overlap receiving by forced = disabled
overlap sending by forced = disabled
protocol side = network
R2 get calling number = none
ISDN virtual connect = disabled
ISDN Layer 2 is DOWN
ISDN Values
ISDN Layer 2 values
k= 7
N200 = 3
N201 = 260
T200 = 1 seconds
T203 = 10 seconds
ISDN Layer 3 values
T301 = 180 seconds
T303 = 4 seconds
T304 = 20 seconds
T305 = 30 seconds
T306 = 30 seconds
T308 = 4 seconds
T310 = 10 seconds
T313 = 10 seconds
T316 = 120 seconds
T322 = 4 seconds
T309 = 90 seconds
N303 = 1
---
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
signalling=mfcr2
mfcr2_variant=mx
trunkgroup = 1,16
spanmap = 1,1,1
[channels]
signalling=mfcr2
mfcr2_variant=mx
context=default
signalling=mfcr2
mfcr2_variant=mx
signalling=mfcr2
mfcr2_variant=mx
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
callerid = asreceived
useincomingcalleridondahditransfer = yes
tonezone = 0 ; 0 is US
channel = 1-15,17-31
signalling=mfcr2
mfcr2_variant=itu
mfcr2_max_ani=7
mfcr2_max_dnis=8
mfcr2_get_ani_first=no
mfcr2_category=national_subscriber
mfcr2_logdir=span1
mfcr2_logging=all
;EOF
cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Tue Dec 22 01:59:02 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED
span=1,1,0,cas,hdb3,crc4
# termtype: te
#bchan=1-15,17-31
#dchan=16
cas=1-15:1101
dchan=16
cas=17-31:1101
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED
span=3,3,0,ccs,hdb3,crc4
# termtype: te
bchan=63-77,79-93
dchan=78
echocanceller=mg2,63-77,79-93
# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED
span=4,4,0,ccs,hdb3,crc4
# termtype: te
bchan=94-108,110-124
dchan=109
echocanceller=mg2,94-108,110-124
# Global data
loadzone= us
defaultzone = us
[default]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN})
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Re: [asterisk-users] Showing name of extension when calling
Magnus Benngård wrote: Is it possible, when placing a call that u see the name of the extension in your diplay? For example, 2 sip.conf entries: [971] callerid=Stefan971 [975] callerid=Magnus975 975 calls 971 today 975 sees 971 in the display but would like to se: Stefan 975 or just Stefan or... This is called Connected Party information display, and it will be in Asterisk 1.8. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Showing name of extension when calling
Kevin P. Fleming wrote: This is called Connected Party information display, and it will be in Asterisk 1.8. Wasn't this scheduled for 1.6.2? Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Release Time Frames
Greetings, Asterisk 1.6.2.0 was released last week. It's time to revisit release plans for both current and future Asterisk releases. For the past few months, there have been discussions regarding some updates to Asterisk release policies. You can find my original -dev list post on this topic here [1]. I also spoke about this as part of my presentation at AstriCon, which you can find a text version of on the Asterisk project blog [2]. After much positive feedback, we have proceeded with implementing these policy updates. I made some additional comments on this topic and noted plans for next major release yesterday [3]. The key things to note are that all current Asterisk releases now have a specified end of life date. Future releases will have EOL dates from their initial release. For additional details regarding maintenance time frames for Asterisk releases, please see the project web site [4]. Thank you all very much for your continued support of Asterisk! [1] http://lists.digium.com/pipermail/asterisk-dev/2009-October/040082.html [2] http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ [3] http://lists.digium.com/pipermail/asterisk-dev/2009-December/041336.html [4] http://www.asterisk.org/asterisk-versions -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call queue with external numbers??
Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1112 1113. What i want to know is, is there a way for me to create such group in asterisk and add that external extension numbers which should be dialed by order, or by 3 rings at a time etcso that i can create that operator group on asterisk side also. PS: I can call real extensions on ericsson without a problem. ___ -- 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] Inquiry:Installing Asterisk 1.4 on CentOS 5.2?
Dan Journo wrote:
I recommend you follow the detailed install guide in this book and
install all the required support programs etc.
http://downloads.oreilly.com/books/9780596510480.pdf
*Thank you for contacting Kesher Communications Ltd.*
*IT Maintenance Clients can now receive a faster response by using our
Live Chat and Support Service: *Click Here
http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508C6B61A91700343lang=ensurpre=PreSurvey**
This email and any files transmitted with it are confidential and
intended solely for the recipient(s). If you are not the named
addressee you should not disseminate, copy or alter this email. Under
no circumstances may this email be distributed without written
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reasonable precautions to ensure no viruses are present in this email,
the company cannot accept responsibility for any loss or damage
arising from the use of this email or attachments. All prices exclude
VAT unless otherwise stated. No responsibility is taken for any
recommendations made by a sender or by Kesher Communications Ltd.
Recipient(s) takes responsibility for any actions taken as a result of
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*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi
motamedi
*Sent:* 22 December 2009 10:47
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on
CentOS 5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
mailto:wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any
errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com
mailto:motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby
wcse...@selbytech.com mailto:wcse...@selbytech.comwrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi
motamed...@gmail.com mailto:motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make
install :
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command not found
The configure script must be executed before running 'make'.
Please run ./configure.
make: *** [makeopts] Error 1
And did you run ./configure like the error message says?
--
Thanks,
--Warren Selby
http://www.selbytech.com http://www.selbytech.com/
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Yes , I did .
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Please find below the output of ./configure :
[r...@mss-0 asterisk-1.4.26]# ./configure
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to
[asterisk-users] Account Code Inbound
I am trying to track inbound and outbound calls by user. In sip.conf, I can add an account code so that all outbound calls from user1 have that as the accountcode in CDR, so that works fine. For inbound, if someone calls user1 direct, I can set the account code in the dial plan like this and it works fine: exten = 700,1,Set(CDR(accountcode)=BOB) exten = 700,2,Dial(SIP/BOB) The problem is if a calls comes in and rings several phones at once, there is no way to set the account code for each user that I can see. Does anybody have any idea on how to do this? Here is an example: exten = 799,1,Dial(SIP/BOBSIP/MARYSIP/DAVESIP/TOM) To get the data I could just search by destination number of BOB, but in the example above the destination number in the CDR is 799, not BOB since that is the number called. I could also search dstchannel as that will show what I want, but I was hoping there was a more generic way to do it (my dialplan is a lot more complex than that listed above and each user has 3-4 lines like 799-BOB, 798-BOB, 797-BOB, so a query to find all of the calls for BOB gets ugly very quickly). Peder ___ -- 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] E1 R2 Congestion Status
On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab kche...@xplorium.comwrote: I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco E1 R2 protocolàsip Gw Find below my error and configuration ,where are the errors in my configuration ? Typically you will be better off in the asterisk-r2 mailing list. Your message is more easily spotted by R2 people. First thing is to check you have green status in your E1, do you? and make sure you have the right clock settings, I never configured a cisco but it seems you configured the cisco to be a slave and I don't see any port in DAHDI system.conf with master clock settings. (span=1,0,0...etc) After that, enable R2 call logging using mfcr2_call_files=yes and pastebin the generated call file (if any), if no file is generated it means the problem is not at R2 but in your local trunk settings (may be dialing in the wrong group or something). -- Moises Silva Senior Software Engineer 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] TDM 400 hardware(?) issue
the machine will lock up because the TDM board or the Dahdi driver goes south. /var/log/messages starts filling up with repeated messages: kernel: TDM PCI Master abort Thank you to everyone who has taken the time to reply. First I am going to apply the fix what you know is broken principle. I have been having issues with the display as well. Since this machine is primarily a server, and the only thing I use the console for is Amarok (it is the house music player), the fact that the text consoles stop working has never been a big issue. CTRL-ALT-F2 produces a blank screen, but the console is really there because I can log in as root and type commands that do get executed, although it's a bit tricky when I can't see what I'm typing. So something is wrong with the display. The video card is a PCI-E card. I don't know how directly the E bus connects to the regular PCI bus, but it's entirely possible that a flaky video card is the whole problem. So I replaced it. Too early to tell if that helps; if I go a few days without any more issues, I'll know that was it. It may not fix the issue, but it was easy and cheap and it needed to be tried anyway for other reasons. This is the price I pay for running a home system where cost is a major issue. I can't afford to have special-purpose servers one for each use; not only is money an issue, but also space and power, so my servers fulfill multiple functions. I realize this is not ideal, and that it would be much better to have the asterisk server dedicated to nothing else, but that isn't a realistic option for me. Up until recently it has always been rock solid. Steve Totaro stot...@totarotechnologies.com wrote: In light of your budget issues, I would switch to quality SIP provider and have my numbers ported. That is a possibility, but finding a quality SIP provider is one issue. I use Teliax over IAX as a backup but have never gotten caller ID to work properly and the sound quality just isn't as good as my POTS line. The other issue is that this puts all my eggs in the Comcast basket. I am reluctant to remove my Qwest land line and have everything depend on Comcast. This also makes it impossible to use the phones without the server machine being up (and asterisk being up), also undesirable (although using the cell phones in that situation is also a possibility). But it is an option. It is a more attractive option if the alternative turns out to be replacing expensive hardware. Lastly there is the WAF (Wife Acceptance Factor), that always has to be considered for home projects. She is the one who doesn't like the sound quality of Teliax. Another provider MIGHT be better but I have no way of knowing, and she won't be happy knowing that we don't have an easy workaround for the house phones if asterisk is down. So this provides another motivation to keep the POTS line if I can do it without a major expense. As far as she is concerned, we don't need asterisk. I like it because I'm a geek and it's cool, and she's OK with that as long as it works. She does enjoy having her messages e-mailed to her, having separate mailboxes, etc., so she does understand the value of these geek projects of mine (MythTV is cool too, a single button press to miss all the annoying commercials), but first and foremost it has to be reliable and at least fulfill the basic functionality. But the WAF has certainly been dropping lately due to all the problems I am having. Other options are going back to old versions of Asterisk. What version are you running? I am already running 1.4 because I have encountered this bug with 1.6: https://issues.asterisk.org/view.php?id=15129 That pretty much prevents inbound calls from working, so I have already had to go back to 1.4 . I am using the asterisk14-1.4.26.3-87 version from ATrpms. I have thought about trying 1.6 with the old zaptel drivers, but that isn't any better as a workaround than what I am already doing, so I haven't gotten around to trying it yet. Darrick Hartman dhart...@djhsolutions.com wrote: Why don't you contact Digium tech support? I have hesitated to do that because my card is fairly old now, but I would certainly do that before permanently abandoning my POTS line or replacing expensive hardware. Steve Tatoro and Bruce Nik recommended Sangoma cards. That is only important if it comes down to replacing the hardware at cost. I am hoping to avoid that. Tilghman Lesher tles...@digium.com wrote: You could try purchasing just the base TDM410 card and move your old modules over from the old card to the new. A little looking around has revealed somebody selling a like new card for $139: Thank you for that information; I didn't know that was an option. My fear with something like this is that if there is a hardware problem that necessitates replacing the card, I don't really have any way of knowing if the problem is in the base card or in one of the modules, so I might end up doing a lot of work and not
Re: [asterisk-users] anonymous calls code
Huge thanks for mentioning what type of channel you are using.
On Tue, Dec 22, 2009 at 5:11 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi C F,
I solved the problem!! It was under my nose...
If you are interested the solution is here:
http://www.misdn.org/index.php/FAQ_chan_mISDN
The right section is: key pad elements
Giorgio Incantalupo
C F wrote:
You would have to create a dialplan for it.
If your provider expects *67 (which is the case here with I/CLEC POTS)
then you would create something like:
exten = _*67[2-9]XX,1,Dial(Zap/g1/${EXTEN})
In the case of PRI you would use:
exten = _*67[2-9]XX,1,SetCallerPres(prohib)
exten = _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3})
On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
does anybody know how to make on-demand anonymous calls? I've tried code
*67# before the number to call but it is working with some providers only.
Any hints?
Thank you.
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Re: [asterisk-users] asterisk x-lite
serach the option en sip.conf:
externip = you public ip
localnet=tus direcciones locales (address local)
saludos
Roman
On Tue, Dec 22, 2009 at 4:26 AM, zehra yildiz zyildi...@gmail.com wrote:
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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Re: [asterisk-users] call queue with external numbers??
At 5:01 PM on 22 Dec 2009, Oguzhan Kayhan wrote: Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1112 1113. What i want to know is, is there a way for me to create such group in asterisk and add that external extension numbers which should be dialed by order, or by 3 rings at a time etcso that i can create that operator group on asterisk side also. PS: I can call real extensions on ericsson without a problem. How about this: exten = 0,1,Dial(DAHDI/G1/,18) exten = 0,n,Dial(DAHDI/G1/1112,18) exten = 0,n,Dial(DAHDI/G1/1113,18) ...where DAHDI/G1 is the PRI connected to the ericsson (group=1 in chan_dahdi.conf), and 18 seconds is 3 rings. You might be able to use Queue(), but I'm not sure if you can add a hunt group and external number as a queue member--you might have to use the Local channel for that. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- 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] Showing name of extension when calling
Doug Lytle wrote: Kevin P. Fleming wrote: This is called Connected Party information display, and it will be in Asterisk 1.8. Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Showing name of extension when calling
On 12/22/09 6:00 PM, Kevin P. Fleming wrote: Doug Lytle wrote: Kevin P. Fleming wrote: This is called Connected Party information display, and it will be in Asterisk 1.8. Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Showing name of extension when calling
On 12/23/09 12:23, Russell Bryant wrote: Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. Wouldn't that imply that it will be in 1.6.3? ___ -- 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] Session Refresh or Codec change
Hi, How asterisk distinguish whether the re-invite is for codec change or for a session refresh? I know that it checks the session version and decides the same. But even if session version is different from the initial invite and but it has the same codec, asterisk identifies that it is a session refresh and does not pass this INVITE to the other end. Whether asterisk compares complete SDP? Is it not a burden on the system? Regards, Prashant MurthyDisclaimer : This message is proprietary to Smartlink Network Systems Limited and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. Smartlink Network Systems Ltd. accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus. __ This email has been scrubbed for your protection by SecureMX. For more information visit http://securemx.in __ ___ -- 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] E1 R2 Congestion Status
Hello,
You are dialing 00223344 with what you show: DAHDI/g1/00223344
That is not a real PSTN number in any country as for as I know. Do you have
the proper outbound route setup? Is your outbound route stripping digits?!
-Bruce
On Tue, Dec 22, 2009 at 9:08 AM, Khaled W Chehab kche...@xplorium.comwrote:
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk –Digium E1 R2 ProtocolàCisco
E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
configuration ?
=
Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on
rev-212-98-156-56 (pid = 3614)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [00223...@default:1] Dial(SIP/98.34.56.216-000e,
DAHDI/g1/00223344) in new stack
[Dec 22 06:02:49] WARNING[4756]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/98.34.56.216-000e' status is
'CONGESTION'
Cisco Gateway
show controller
Controller E1 slot(0)/port(0)
E1 Link is UP
No Alarm detected.
Applique type is Channelized E1.
Framing is CRC4, Line Code is HDB3.
Signalling type is R2-MFC.
0 Line Code Violations, 0 Framing Bit Errors
0 Far End Block Errors, 0 CRC Errors
signalling type = r2
clock source = slave
channel group 0 = 1-31
1 2 3
allocated timeslots = YYYNYYY
outgoing barred channel group =
channel order = ascending
b-channel negotiation = exclusive
overlap receiving by forced = disabled
overlap sending by forced = disabled
protocol side = network
R2 get calling number = none
ISDN virtual connect = disabled
ISDN Layer 2 is DOWN
ISDN Values
ISDN Layer 2 values
k= 7
N200 = 3
N201 = 260
T200 = 1 seconds
T203 = 10 seconds
ISDN Layer 3 values
T301 = 180 seconds
T303 = 4 seconds
T304 = 20 seconds
T305 = 30 seconds
T306 = 30 seconds
T308 = 4 seconds
T310 = 10 seconds
T313 = 10 seconds
T316 = 120 seconds
T322 = 4 seconds
T309 = 90 seconds
N303 = 1
---
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
signalling=mfcr2
mfcr2_variant=mx
trunkgroup = 1,16
spanmap = 1,1,1
[channels]
signalling=mfcr2
mfcr2_variant=mx
context=default
signalling=mfcr2
mfcr2_variant=mx
signalling=mfcr2
mfcr2_variant=mx
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
callerid = asreceived
useincomingcalleridondahditransfer = yes
tonezone = 0 ; 0 is US
channel = 1-15,17-31
signalling=mfcr2
mfcr2_variant=itu
mfcr2_max_ani=7
mfcr2_max_dnis=8
mfcr2_get_ani_first=no
mfcr2_category=national_subscriber
mfcr2_logdir=span1
mfcr2_logging=all
;EOF
cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Tue Dec 22 01:59:02 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED
span=1,1,0,cas,hdb3,crc4
# termtype: te
#bchan=1-15,17-31
#dchan=16
cas=1-15:1101
dchan=16
cas=17-31:1101
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED
span=3,3,0,ccs,hdb3,crc4
# termtype: te
bchan=63-77,79-93
dchan=78
echocanceller=mg2,63-77,79-93
# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED
span=4,4,0,ccs,hdb3,crc4
# termtype: te
bchan=94-108,110-124
dchan=109
echocanceller=mg2,94-108,110-124
# Global data
loadzone= us
defaultzone = us
[default]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN})
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Re: [asterisk-users] Showing name of extension when calling
On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote: On 12/23/09 12:23, Russell Bryant wrote: Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. Wouldn't that imply that it will be in 1.6.3? What was once going to be 1.6.3 is now going to be 1.8. There is no 1.6.3 in the planning stages at this time. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Showing name of extension when calling
why the edition jump from 1.6.2 to 1.8 , what's the reason? the number of the edition always confuse me. 2009/12/23 Tilghman Lesher tles...@digium.com: On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote: On 12/23/09 12:23, Russell Bryant wrote: Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. Wouldn't that imply that it will be in 1.6.3? What was once going to be 1.6.3 is now going to be 1.8. There is no 1.6.3 in the planning stages at this time. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Sucan ___ -- 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] Showing name of extension when calling
On Wednesday 23 December 2009 01:31:04 Zhang Shukun wrote: 2009/12/23 Tilghman Lesher tles...@digium.com: On Tuesday 22 December 2009 23:08:36 Rob Hillis wrote: On 12/23/09 12:23, Russell Bryant wrote: Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. Wouldn't that imply that it will be in 1.6.3? What was once going to be 1.6.3 is now going to be 1.8. There is no 1.6.3 in the planning stages at this time. why the edition jump from 1.6.2 to 1.8 , what's the reason? the number of the edition always confuse me. http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP realm
Can I define the realm on a per peer basis ?? Can I define a realm to be used for one peer and another realm for another peer in sip.conf ?? I have an ITSP that I need to authenticate with a realm that they set. But this realm is not valuable for other peers. Jonas. ___ -- 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
