[asterisk-users] Start asterisk on boot
Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk -f But it doesn't work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- 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] Start asterisk on boot
On Mon, 2009-01-26 at 09:32 +0100, Ralf Träskman wrote: Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk –f But it doesn’t work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P Please consider the environment before printing this e-mail I am using the /etc/init.d/asterisk startup file and works Just make sure that permissions are set correctly. The default asterisk package from ubuntu reps runs as non-root so if you have installed it prior to installing 1.6 some things will be warped. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.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] Start asterisk on boot
2009/1/26 Ralf Träskman r...@adlibris.com Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk –f But it doesn't work. Regards /ralf 1. Copy relevant file from contrib directory into /etc/init.d directory (while renaming it asterisk) 2. Then sudo update-rc.d asterisk defaults and it's done Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P *Please consider the environment before printing this e-mail* ___ -- 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] Start asterisk on boot
How did you installed asterisk? 2009/1/26 Ralf Träskman r...@adlibris.com: Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk –f But it doesn't work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P Please consider the environment before printing this e-mail ___ -- 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 -- مصطفى ب ___ -- 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] Start asterisk on boot
On Mon, Jan 26, 2009 at 09:32:26AM +0100, Ralf Träskman wrote: Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk -f exec /usr/sbin/asterisk -U asterisk please. How do you intend to stop it? -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/24 Matthew Fredrickson cres...@digium.com Olivier wrote: Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME are equipped today with PBX connected with TE/PtMP interfaces to telco BRI lines. When we address those SME, my opinion is that it's very useful to be able to support any combination of TE/NT, PtP/PtMP modes. Latest 1.6 Asterisk and 1.4.8 Libpri introduced a new set of welcomed ISDN BRI features. Unfortunately, NT/PtMP is not available at this time, in latest Zaptel/Asterisk/Libpri. My question is what is the policy concerning NT/PtMP ? Is it really hard to extend Libpri to support this mode ? Or shall mISDN remain the way to go when NT/PtMP is needed ? Hey Olivier, I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. I know how much we owe you for BRI support and lipri in general and I really thank you for that. NT PTMP is very significantly different, in that you have to do much more from a TEI management perspective. Most people's needs that I saw were actually fulfilled in using either NT or TE PTP or TE PTMP, since they were interfacing with PBXs or using TE-PTMP trunks from the telephone network to provide voice trunks for Asterisk. From my point of view, the most important feature is TE-PTMP as this the one used here (in France) when connecting a new Asterisk-based IPBX to ISDN (I really don't know why TE-PTP is not used for that). From the same point of view, 2nd most needed feature is NT-PTMP when connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. I'm afraid this mISDN option is not very encouraging these days : - misdn mailing list is not working these days (I'm hoping I'm wrong but it seems to be the case), - mISDN won't compile on latest 2.6.26 kernels so maybe mISDN developpers are thinking B410P features inclusion in 1.6 sets a mark in Asterisk BRI policy and it's not worth developing mISDN anymore. So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) I don't want to be misunderstood while writing this memo : - a very good job have been made lately in dahdi/libpri with B410P inclusion - if others are thinking NT-PTMP is needed or disagree with that, let them say it here and now, as it might take a long time to integrate this feature Regards Matthew Fredrickson Digium, Inc. ___ -- 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
[asterisk-users] Voicemail
Is there a way to customize the voicemail navigation system? Google shows some discussion of it in late 2006, but I see no references to it being implemented. Many thanks, -Justin ___ -- 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] soft phone
Bayardo Sanchez wrote: i am used eyebeam is a good softphone On Sun, Jan 25, 2009 at 12:55 PM, David fire ddf...@gmail.com mailto:ddf...@gmail.com wrote: hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? Zoiper biz can do that, when used with a provisioning server. thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] soft phone
bilal ghayyad wrote: But how can I have a one that has messanger facilities? Ability to see who is login and send sms, in addition to do a calls? Zoiper communicator can do that, but there is no stable release yet and its probably not user friendly enough yet, but will be in a couple of weeks (http://beta.zoiper.com) Regards Bilal --- Try iaxLite or sipLite - Original Message - From: David fire To: bilmar...@yahoo.com ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 26, 2009 7:43 AM Subject: Re: [asterisk-users] soft phone there isnt any free soft phone wich support G729. 2009/1/25 bilal ghayyad bilmar...@yahoo.com But eyebeam support g729 codec? And does it support IAX? By the way: I am looking for softphone to has as messanger? Ability to have users login, and sending text message in addition to having voice calls? Any advise? Where to be find? Regards Bilal -- i am used eyebeam is a good softphone On Sun, Jan 25, 2009 at 12:55 PM, David fire ddf...@gmail.com wrote: hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks ___ -- 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
[asterisk-users] custom cdr userfiled
Dear, I added new field to cdr table , named service and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten = _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) As a quick alternative look at zaphfc and friends, but don't expect it to be trouble-free. -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.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] Start asterisk on boot
Hi That didnt work either, do i have to set some permissions? /ralf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: den 26 januari 2009 09:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Start asterisk on boot 2009/1/26 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk -f But it doesn't work. Regards /ralf 1. Copy relevant file from contrib directory into /etc/init.d directory (while renaming it asterisk) 2. Then sudo update-rc.d asterisk defaults and it's done Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- 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] Voicemail
Justin Fletcher schrieb: Is there a way to customize the voicemail navigation system? Google shows some discussion of it in late 2006, but I see no references to it being implemented. In Asterisk 1.6 there's MiniVM (don't be mislead by mini). If you read German have a look at http://www.das-asterisk-buch.de/2.1/minivm.html Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] custom cdr userfiled
Pezhman Lali schrieb: I added new field to cdr table , named service and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten = _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? Depends on the CDR backend. custom_odbc: yes. mysql: no. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] custom cdr userfiled
so great, I am using odbc . but I can not find custom_odbc . is any keywords or links to useful documents? best Mani --- On Mon, 1/26/09, Philipp Kempgen philipp.kemp...@amooma.de wrote: From: Philipp Kempgen philipp.kemp...@amooma.de Subject: Re: [asterisk-users] custom cdr userfiled To: Asterisk Users asterisk-users@lists.digium.com Date: Monday, January 26, 2009, 1:18 PM Pezhman Lali schrieb: I added new field to cdr table , named service and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten = _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? Depends on the CDR backend. custom_odbc: yes. mysql: no. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Voicemail
On Mon, Jan 26, 2009 at 10:39 AM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Justin Fletcher schrieb: Is there a way to customize the voicemail navigation system? Google shows some discussion of it in late 2006, but I see no references to it being implemented. In Asterisk 1.6 there's MiniVM (don't be mislead by mini). If you read German have a look at http://www.das-asterisk-buch.de/2.1/minivm.html Philipp Kempgen I don't read German, but from what I can piece together plus the config examples it looks pretty straight forward. Thanks! -Justin ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/26 Stelios Koroneos skoron...@digital-opsis.com So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) As a quick alternative look at zaphfc and friends, but don't expect it to be trouble-free. Excuse me but I'm not sure I'm getting it. What I had in mind is : ISDN -BRI Asterisk with PCI board BRI --- Legacy PBX For that, in countries where PTMP is the norm (is it ?) for small PBX (what is a small PBX, I don't know), you both need TE PTMP and NT PTMP. Options are Bristuff, mISDN, Dahdi (which now support BRI but NT PTMP is missing at this time) Alternative is use a box that support those 2 modes such as : ISDN -BRI Patton box BRI --- Legacy PBX | | Asterisk So, Stelios, what is the zaphfc and friends you were referring to ? By the way, do you mostly use TE PTMP or TE PTP ? Regards -- Stelios S. Koroneos Digital OPSiS - Embedded Intelligence Tel +30 210 9858296 Ext 100 Fax +30 210 9858298 http://www.digital-opsis.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 ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
On Mon, Jan 26, 2009 at 11:09:35AM +0100, Olivier wrote: 2009/1/26 Stelios Koroneos skoron...@digital-opsis.com So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) As a quick alternative look at zaphfc and friends, but don't expect it to be trouble-free. Excuse me but I'm not sure I'm getting it. What I had in mind is : ISDN -BRI Asterisk with PCI board BRI --- Legacy PBX For that, in countries where PTMP is the norm (is it ?) for small PBX (what is a small PBX, I don't know), you both need TE PTMP and NT PTMP. Options are Bristuff, mISDN, Dahdi (which now support BRI but NT PTMP is missing at this time) Alternative is use a box that support those 2 modes such as : ISDN -BRI Patton box BRI --- Legacy PBX | | Asterisk So, Stelios, what is the zaphfc and friends you were referring to ? By the way, do you mostly use TE PTMP or TE PTP ? If you connect to another PBX, why would you use PTMP? -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Olivier a écrit : From my point of view, the most important feature is TE-PTMP as this the one used here (in France) when connecting a new Asterisk-based IPBX to ISDN (I really don't know why TE-PTP is not used for that). Well this may depend of some parameter. We have a dual BRI line from France Telecom, in PTP mode. From the same point of view, 2nd most needed feature is NT-PTMP when connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Well, i agree on something, when you are using PTMP mode, being able to do it both way is very usefull (for transperency placing an * between a legacy pabx and the 'outside world'). However it's when their is at least one NT mode, you could always reconfigure the pabx (but you lose the ability to reconnect him directly to the bri line in case of problem) Please, do not hesitate to comment. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. I'm afraid this mISDN option is not very encouraging these days : - misdn mailing list is not working these days (I'm hoping I'm wrong but it seems to be the case), - mISDN won't compile on latest 2.6.26 kernels so maybe mISDN developpers are thinking B410P features inclusion in 1.6 sets a mark in Asterisk BRI policy and it's not worth developing mISDN anymore. NT PTMP is very significantly different, in that you have to do much Their is an unannonced mISN-1.1.9 release (and even 1.1.9.1 now). Since their is no release note i can't say for sur they fixed this but this may be worth a try So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) Why so ? As for myself am very interested in red-fone boxes, being able to separate the ipbx and the PRI 'access point' ahs some advantage (easier resiliency) ___ -- 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] Ntework Card
lspci Returns me below Output :_ [r...@vicidialnow ~]# lspci 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Gr aphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connectio n (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contolle r #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (r ev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (r ev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (r ev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controll er (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Co ntroller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Co ntroller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) ifconfig -a Returns : _ ifconfig -a eth0 Link encap:Ethernet HWaddr 00:1E:58:A7:3D:73 inet addr:121.247.xxx.xxx Bcast:121.247.xxx.xxx Mask:255.255.255.0 inet6 addr: fe80::21e:58ff:fea7:3d73/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:415 errors:0 dropped:0 overruns:0 frame:0 TX packets:478 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:51514 (50.3 KiB) TX bytes:95169 (92.9 KiB) Interrupt:169 loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:112962 errors:0 dropped:0 overruns:0 frame:0 TX packets:112962 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:9165137 (8.7 MiB) TX bytes:9165137 (8.7 MiB) sit0 Link encap:IPv6-in-IPv4 NOARP MTU:1480 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) On Sun, Jan 25, 2009 at 2:53 PM, David @ULC ucoms2...@gmail.com wrote: *Quote:* [r...@vicidialnow src]# service network restart Shutting down interface eth0: [ OK ] Shutting down loopback interface: [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: [ OK ] Bringing up interface eth1: skge device eth1 does not seem to be present, delaying initialization. [FAILED] [r...@vicidialnow src]# eth1 is the Oboard card. I did install driver but its same. I am sure Driver is a correct one as after installing driver it showed me the card which is eth1 but due to soem issues I deleted that network interface from webmin. But now its NOT detecting. Kindly advice ___ -- 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] Ntework Card
I also installed the driver : [r...@vicidialnow ~]# tar zxf e1000-8.0.6.tar.gz [r...@vicidialnow ~]# cd e1000-8.0.6/src/ [r...@vicidialnow src]# make install make -C /lib/modules/2.6.18-53.el5/build SUBDIRS=/root/e1000-8.0.6/src modules make[1]: Entering directory `/usr/src/kernels/2.6.18-53.el5-i686' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/kernels/2.6.18-53.el5-i686' # remove all old versions of the driver find /lib/modules/2.6.18-53.el5 -name e1000.ko -exec rm -f {} \; || true find /lib/modules/2.6.18-53.el5 -name e1000.ko.gz -exec rm -f {} \; || true install -D -m 644 e1000.ko /lib/modules/2.6.18-53.el5/kernel/drivers/net/e1000/e1000.ko /sbin/depmod -a || true install -D -m 644 e1000.7.gz /usr/share/man/man7/e1000.7.gz man -c -P'cat /dev/null' e1000 || true On Mon, Jan 26, 2009 at 4:07 PM, David @ULC ucoms2...@gmail.com wrote: lspci Returns me below Output :_ [r...@vicidialnow ~]# lspci 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Gr aphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connectio n (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contolle r #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (r ev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (r ev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (r ev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controll er (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Co ntroller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Co ntroller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) ifconfig -a Returns : _ ifconfig -a eth0 Link encap:Ethernet HWaddr 00:1E:58:A7:3D:73 inet addr:121.247.xxx.xxx Bcast:121.247.xxx.xxx Mask:255.255.255.0 inet6 addr: fe80::21e:58ff:fea7:3d73/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:415 errors:0 dropped:0 overruns:0 frame:0 TX packets:478 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:51514 (50.3 KiB) TX bytes:95169 (92.9 KiB) Interrupt:169 loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:112962 errors:0 dropped:0 overruns:0 frame:0 TX packets:112962 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:9165137 (8.7 MiB) TX bytes:9165137 (8.7 MiB) sit0 Link encap:IPv6-in-IPv4 NOARP MTU:1480 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) On Sun, Jan 25, 2009 at 2:53 PM, David @ULC ucoms2...@gmail.com wrote: *Quote:* [r...@vicidialnow src]# service network restart Shutting down interface eth0: [ OK ] Shutting down loopback interface: [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: [ OK ] Bringing up interface eth1: skge device eth1 does not seem to be present, delaying
[asterisk-users] German date format in voicemail emails
Hi! I want to configure voicemail to send emails with the date of the message in German/Austria, that means: Montag, 26 Jänner 2009 instead of Monday, 26 January 2009 voicemail.conf refers to man strftime. This refers to the current locales. So, I tried export LANG=de export LC_ALL=de_DE before starting Asterisk. Unfortunately the date format is still in English. Any hints how to achieve the German language date format? thanks klaus ___ -- 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] Vicidialnow
where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.comwrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] German date format in voicemail emails
Klaus Darilion schrieb: I want to configure voicemail to send emails with the date of the message in German/Austria, that means: Montag, 26 Jänner 2009 instead of Monday, 26 January 2009 voicemail.conf refers to man strftime. This refers to the current locales. So, I tried export LANG=de export LC_ALL=de_DE before starting Asterisk. Unfortunately the date format is still in English. Any hints how to achieve the German language date format? Do you have these locales installed? And what's the default locale on your system? Debian: aptitude install locales dpkg-reconfigure locales (Locales to be generated: check the ones you want to install) (Default locale for the system environment: e.g. de_DE.UTF-8) However I'm not sure if that helps. Maybe Asterisk always uses the C locale to expand emaildateformat. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Auto Detect
Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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] Start asterisk on boot
2009/1/26 Ralf Träskman r...@adlibris.com Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description Asterisk daemon start on runlevel-2 stop on shutdown respawn exec //usr/sbin/asterisk –f But it doesn't work. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P *Please consider the environment before printing this e-mail* ___ -- 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 go to the source directory where you did ./configure make make install and run make config this will make all the start/stop scripts for you. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Digium TE220 card partially detected
Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 .. Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware pci::03:08.0 wct4xxp+ d161:0220 Wildcard TE220 (4th Gen) astpbx ~ # But dahdi_tool does not show the card and dahdi_cfg gives DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) System info is provided here - http://asteriskpbx.ru/pastebin/12 Please advise. -- Maxim Litnitskiy http://asteriskpbx.ru http://asterisk-support.ru ___ -- 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] Vicidialnow
the vicidial proyect is alive? the last stable releace is from 2007 thanks David 2009/1/26 David fire ddf...@gmail.com where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.comwrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.comwrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Auto Detect
On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk Hardware of what type? For Zaptel/DAHDI: zapconf / dahdi_genconf . -- 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] Digium TE220 card partially detected
On Mon, Jan 26, 2009 at 03:08:13PM +0300, Maxim Litnitskiy wrote: Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 .. Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware pci::03:08.0 wct4xxp+ d161:0220 Wildcard TE220 (4th Gen) Fine. The card is indeed handled by the module (+). astpbx ~ # But dahdi_tool does not show the card and dahdi_cfg gives DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) Next: are you sure that it is configured for E1 and not for T1? What is the output of: cat /proc/dahdi/1 System info is provided here - http://asteriskpbx.ru/pastebin/12 -- 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] Auto Detect
Tzafrir Cohen schrieb: On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote: Which command to run which will auto detect all hardwares present in the system ? Hardware of what type? The OP clearly said *all* hardware. :-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 caused Kernel to segfault
someone change the top posting complains emails to sarcastic mails??? David 2009/1/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Sun, Jan 25, 2009 at 08:59:16PM -0200, David fire wrote: you dont have dahdi properly installed. did you install BOTH dahdi utils and dahdi modules Wow, you finally realized that? What is the output of: uname -r find /lib/modules -name dahdi.ko -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Network Card
Sorry, but any further help withhttp://lists.digium.com/pipermail/asterisk-users/2009-January/225548.html http://lists.digium.com/pipermail/asterisk-users/2009-January/225548.html ___ -- 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] Ntework Card
[r...@vicidialnow ~]# cat /etc/sysconfig/network-scripts/ifcfg-eth1 NAME= BOOTPROTO=none TYPE=Ethernet DEVICE=eth1 MTU= NETMASK=255.255.255.0 BROADCAST=192.168.0.100 IPADDR=192.168.0.100 NETWORK=192.168.0.0 ONBOOT=no On Mon, Jan 26, 2009 at 4:07 PM, David @ULC ucoms2...@gmail.com wrote: lspci Returns me below Output :_ [r...@vicidialnow ~]# lspci 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Gr aphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connectio n (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contolle r #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (r ev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (r ev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (r ev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controll er (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Co ntroller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Co ntroller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) ifconfig -a Returns : _ ifconfig -a eth0 Link encap:Ethernet HWaddr 00:1E:58:A7:3D:73 inet addr:121.247.xxx.xxx Bcast:121.247.xxx.xxx Mask:255.255.255.0 inet6 addr: fe80::21e:58ff:fea7:3d73/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:415 errors:0 dropped:0 overruns:0 frame:0 TX packets:478 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:51514 (50.3 KiB) TX bytes:95169 (92.9 KiB) Interrupt:169 loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:112962 errors:0 dropped:0 overruns:0 frame:0 TX packets:112962 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:9165137 (8.7 MiB) TX bytes:9165137 (8.7 MiB) sit0 Link encap:IPv6-in-IPv4 NOARP MTU:1480 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) On Sun, Jan 25, 2009 at 2:53 PM, David @ULC ucoms2...@gmail.com wrote: *Quote:* [r...@vicidialnow src]# service network restart Shutting down interface eth0: [ OK ] Shutting down loopback interface: [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: [ OK ] Bringing up interface eth1: skge device eth1 does not seem to be present, delaying initialization. [FAILED] [r...@vicidialnow src]# eth1 is the Oboard card. I did install driver but its same. I am sure Driver is a correct one as after installing driver it showed me the card which is eth1 but due to soem issues I deleted that network interface from webmin. But now its NOT detecting. Kindly advice ___ -- 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] Auto Detect
On Mon, Jan 26, 2009 at 01:45:56PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote: Which command to run which will auto detect all hardwares present in the system ? Hardware of what type? The OP clearly said *all* hardware. :-) Most of your hardware is something a standard linux distro would handle. * chan_alsa and chan_oss will then use whatever you defined through ALSA/OSS (and usually the distro does that for you) * Likewise chan_consoles sees whatever pulseaudio sees, I believe. * chan_vpb usually detects hardware automatically (in libvpb) * chan_phone: no idea * chan_misdn: should have something similar. I'm not familiar with it. * chan_usbradio: I'm likewise completely unfamiliar with that. So I guess I was close enough. -- 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] German date format in voicemail emails
Philipp Kempgen schrieb: Klaus Darilion schrieb: I want to configure voicemail to send emails with the date of the message in German/Austria, that means: Montag, 26 Jänner 2009 instead of Monday, 26 January 2009 voicemail.conf refers to man strftime. This refers to the current locales. So, I tried export LANG=de export LC_ALL=de_DE before starting Asterisk. Unfortunately the date format is still in English. Any hints how to achieve the German language date format? Do you have these locales installed? And what's the default locale on your system? Meanwhile yes :-) Debian: aptitude install locales dpkg-reconfigure locales (Locales to be generated: check the ones you want to install) (Default locale for the system environment: e.g. de_DE.UTF-8) However I'm not sure if that helps. Maybe Asterisk always uses the C locale to expand emaildateformat. I tried setting the locale in app_voicemail manually (ported from minivm) and then it works. But now I try setting it in the environment before starting Asterisk, but that does not work . klaus ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/26 Benoit maver...@maverick.eu.org Olivier a écrit : From my point of view, the most important feature is TE-PTMP as this the one used here (in France) when connecting a new Asterisk-based IPBX to ISDN (I really don't know why TE-PTP is not used for that). Well this may depend of some parameter. We have a dual BRI line from France Telecom, in PTP mode. Yes, also, I really don't know what makes a France Telecom line be configured in PTP or PTMP. Anyway, according your own experience, how frequent is this PTP case ? From the same point of view, 2nd most needed feature is NT-PTMP when connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Well, i agree on something, when you are using PTMP mode, being able to do it both way is very usefull (for transperency placing an * between a legacy pabx and the 'outside world'). However it's when their is at least one NT mode, you could always reconfigure the pabx (but you lose the ability to reconnect him directly to the bri line in case of problem) That's the point : as much as possible, we're trying to avoid any re-configuration of legacy equipment. So, how would you order TE/NT, PtP/PtMP combination starting from the most useful (for yourself) to the least one. For instance, myself, I would classify them like this : 1. TE PtMP (the most needed) 2. NT PtMP 3. TE PtP 4. NT PtP (the least needed) Please, do not hesitate to comment. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. I'm afraid this mISDN option is not very encouraging these days : - misdn mailing list is not working these days (I'm hoping I'm wrong but it seems to be the case), - mISDN won't compile on latest 2.6.26 kernels so maybe mISDN developpers are thinking B410P features inclusion in 1.6 sets a mark in Asterisk BRI policy and it's not worth developing mISDN anymore. NT PTMP is very significantly different, in that you have to do much Their is an unannonced mISN-1.1.9 release (and even 1.1.9.1 now). Since their is no release note i can't say for sur they fixed this but this may be worth a try So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) Why so ? As for myself am very interested in red-fone boxes, being able to separate the ipbx and the PRI 'access point' ahs some advantage (easier resiliency) I agree that separating functions into several boxes is fine but today, it would be touchy (if possible) to build all boxes using Asterisk and Digium boards ... If we want Asterisk ecosystem to develop (buying boards paying software development), it's important to be able to build those boxes using components made by companies contributing to Asterisk code ... ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/26 Benoit maver...@maverick.eu.org Olivier a écrit : From my point of view, the most important feature is TE-PTMP as this the one used here (in France) when connecting a new Asterisk-based IPBX to ISDN (I really don't know why TE-PTP is not used for that). Well this may depend of some parameter. We have a dual BRI line from France Telecom, in PTP mode. From the same point of view, 2nd most needed feature is NT-PTMP when connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Well, i agree on something, when you are using PTMP mode, being able to do it both way is very usefull (for transperency placing an * between a legacy pabx and the 'outside world'). However it's when their is at least one NT mode, you could always reconfigure the pabx (but you lose the ability to reconnect him directly to the bri line in case of problem) Please, do not hesitate to comment. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. I'm afraid this mISDN option is not very encouraging these days : - misdn mailing list is not working these days (I'm hoping I'm wrong but it seems to be the case), - mISDN won't compile on latest 2.6.26 kernels so maybe mISDN developpers are thinking B410P features inclusion in 1.6 sets a mark in Asterisk BRI policy and it's not worth developing mISDN anymore. NT PTMP is very significantly different, in that you have to do much Their is an unannonced mISN-1.1.9 release (and even 1.1.9.1 now). Since their is no release note i can't say for sur they fixed this but this may be worth a try I didn't know that !! Now I can see it appearing : it dates from today ! Thanks for letting us know ... (as mISDN list is silent) So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) Why so ? As for myself am very interested in red-fone boxes, being able to separate the ipbx and the PRI 'access point' ahs some advantage (easier resiliency) I agree that separating functions into several boxes is fine but today, it would be touchy (if possible) to build all boxes using Asterisk and Digium boards, for example ... If we want Asterisk ecosystem to develop (buying boards paying software development), it's important to be able to build those boxes using components made by companies (Digium, Xorcom, ...) contributing to Asterisk code ... ___ -- 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] Auto Detect
try lspci 2009/1/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Jan 26, 2009 at 01:45:56PM +0100, Philipp Kempgen wrote: Tzafrir Cohen schrieb: On Mon, Jan 26, 2009 at 05:24:03PM +0530, David @ULC wrote: Which command to run which will auto detect all hardwares present in the system ? Hardware of what type? The OP clearly said *all* hardware. :-) Most of your hardware is something a standard linux distro would handle. * chan_alsa and chan_oss will then use whatever you defined through ALSA/OSS (and usually the distro does that for you) * Likewise chan_consoles sees whatever pulseaudio sees, I believe. * chan_vpb usually detects hardware automatically (in libvpb) * chan_phone: no idea * chan_misdn: should have something similar. I'm not familiar with it. * chan_usbradio: I'm likewise completely unfamiliar with that. So I guess I was close enough. -- 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 ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Jan 26, 2009 at 11:09:35AM +0100, Olivier wrote: 2009/1/26 Stelios Koroneos skoron...@digital-opsis.com So my opinion is that these NT-PTMP is really and urgently needed, especially if this TEI management is rather complex and therefore would take a long time to develop and stabilize. The alternative is to keep using those Patton, Quintum, etc ... boxes which is not what we would prefer ;-)) As a quick alternative look at zaphfc and friends, but don't expect it to be trouble-free. Excuse me but I'm not sure I'm getting it. What I had in mind is : ISDN -BRI Asterisk with PCI board BRI --- Legacy PBX For that, in countries where PTMP is the norm (is it ?) for small PBX (what is a small PBX, I don't know), you both need TE PTMP and NT PTMP. Options are Bristuff, mISDN, Dahdi (which now support BRI but NT PTMP is missing at this time) Alternative is use a box that support those 2 modes such as : ISDN -BRI Patton box BRI --- Legacy PBX | | Asterisk So, Stelios, what is the zaphfc and friends you were referring to ? By the way, do you mostly use TE PTMP or TE PTP ? If you connect to another PBX, why would you use PTMP? Because the other PBX was connected in TE PTMP mode to ISDN lines and you need to avoid any configuration change ... -- 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 ___ -- 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] Auto Detect
lspci ~ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contoller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) but *service network restart RETURNS me* *** Shutting down interface eth0: [ OK ]** Shutting down loopback interface: [ OK ]** Bringing up loopback interface: [ OK ]** Bringing up interface eth0: [ OK ]** Bringing up interface eth1: skge device eth1 does not seem to be present,** delaying initialization.** [FAILED]* On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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] Auto Detect
you have asked several questions that have little or nothing to do with asterisk. Perhaps you should purchase some consulting time from a linux admin, join your local Linux Users Group, or at least ask your questions in a newbies forum for the version of linux you have chosen... If you insist on doing this yourself, recognize that you will get back only what you're willing to put in. Please consider going to a good public library and reading a general guide to getting started with linux. If your local public library doesn't have these titles, ask about interlibrary loans with the librarian on duty. I recommend O'Reilly books, specifically: http://oreilly.com/catalog/9780596100292/ When you have a better understanding of unix and linux, I recommend you browse through the O'Reilly guide to Asterisk, referred to by me as the starfish book: http://oreilly.com/catalog/9780596510480/index.html Now regarding hardware, various linux distributions handle this differently. CentOS and other flavors of linux generally support the command: /sbin/lspci you will need to become root, and type lspci and this will list much of the hardware in your system, but it won't list USB devices and some other peripherals like printers. If you're coming from the Windows world, there's no such thing as Control Panel's hardware list. If this is for a business and you're working on a short deadline, you're going to find this process very frustrating and you should hire somebody yesterday rather than trying to aggregate the years of linux knowledge, as well as asterisk-specific know-how that people like me have accumulated. If you want the easy way out and know how to click things, you may find joy with projects including: webmin FreePBX However, webmin only gets you most of the way to administering your linux machine, and FreePBX will get you started with asterisk but will give you just enough rope to hang yourself once you try to do something custom and complicated. If you need to write very custom, business-critical apps, you need a consultant or a lot of patience to bring yourself up-to-speed. Why did you pick CentOS? CentOS is not a good place for a new linux user to start. I'm not convinced you made an informed decision, and I'm thinking you should start over again with Ubuntu, which is more user-friendly for a beginner, and has several Ubuntu-specific getting started manuals at any self-respecting big-box book seller, like BN and Borders. Ubuntu also has excellent forums for new linux users. Good luck, and welcome back when you have enough of a handle on your linux administration to move on to configuring asterisk. On Mon, Jan 26, 2009 at 6:54 AM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Tzafrir, Personally, how would you classify most needed BRI modes, from most valuable to least valuable (see other posts in this thread) ? ___ -- 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] Auto Detect
You seem to have two network devices in that system: intel, probably onboard dlink, probably a pci or pci-express card one is working fine, the other is not. This is usually a kernel driver problem you will need to work out. Check out the CentOS forums to see whether your dlink card is supported by your version of CentOS. Read up on kernel modules, the commands for seeing which ones you're using, how you use other kernel modules, and which ones may or may not support the network devices you are trying to use. I've never tried to use your dlink card and don't know whether it will work with linux. Or take my suggestion and start over again with Ubuntu, and then use the Ubuntu forums. On Mon, Jan 26, 2009 at 8:16 AM, David @ULC ucoms2...@gmail.com wrote: lspci ~ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contoller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) but service network restart RETURNS me Shutting down interface eth0: [ OK ] Shutting down loopback interface: [ OK ] Bringing up loopback interface: [ OK ] Bringing up interface eth0: [ OK ] Bringing up interface eth1: skge device eth1 does not seem to be present, delaying initialization. [FAILED] On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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] Auto Detect
Probably not much use to you in this instance, but its always nice to know that lspci -v prints the kernel driver in use and kernel modules. e.g. 10:00.0 Ethernet controller: Broadcom Corporation NetLink BCM5787M Gigabit Ethernet PCI Express (rev 02) Subsystem: Hewlett-Packard Company Device 30c2 Flags: bus master, fast devsel, latency 0, IRQ 220 Memory at d000 (64-bit, non-prefetchable) [size=64K] Expansion ROM at ignored [disabled] Capabilities: access denied Kernel driver in use: tg3 Kernel modules: tg3 Bails David @ULC wrote: lspci ~ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contoller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) but *service network restart RETURNS me* *** Shutting down interface eth0: [ OK ]** Shutting down loopback interface: [ OK ]** Bringing up loopback interface: [ OK ]** Bringing up interface eth0: [ OK ]** Bringing up interface eth1: skge device eth1 does not seem to be present,** delaying initialization.** [FAILED]* On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] * Queues with legacy pbx extensions ?
Hello Everybody I am using Trixbox 2.4 (with TE420P PRI lines) .. my setup is like Calls --Asterisk--legacy pbx---analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents like 102,103,104 are present. Now my question is (might be very silly) : 1. I dont have a queue stats software for my legacy pbx, the agents stil logon and logout on their legacy pbx..I was wondering if its possible that all these 102,103,104 etc login to a queue which resides on asterisk ? Can they login and logout of the asterisk queue..I am asking this since there are very nice asterisk queue reporting utilities out there for free..and i wud like to use them ...this will also help me using screen pop-ups for my agents...i know SIP extensions can simplify my setup but echo problems scare me also i dont want to throw my legacy pbx on which i invested heavily.. Can anyone throw some pointers ? Thanks in advance Sriram ___ -- 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] Auto Detect
lspci -v RETURNS me : ___ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Gr aphics Controller (rev 03) (prog-if 00 [VGA]) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at e020 (32-bit, non-prefetchable) [size=1M] Memory at d000 (64-bit, prefetchable) [size=256M] I/O ports at 3440 [size=8] Capabilities: [90] Message Signalled Interrupts: 64bit- Queue=0/0 Enable - Capabilities: [d0] Power Management version 2 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, fast devsel, latency 0 Memory at e010 (32-bit, non-prefetchable) [size=1M] Capabilities: [d0] Power Management version 2 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connectio n (rev 02) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, fast devsel, latency 0, IRQ 9 Memory at e030 (32-bit, non-prefetchable) [size=128K] Memory at e0324000 (32-bit, non-prefetchable) [size=4K] I/O ports at 30c0 [size=32] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable - 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contolle r #4 (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, medium devsel, latency 0, IRQ 177 I/O ports at 30a0 [size=32] 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #5 (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, medium devsel, latency 0, IRQ 225 I/O ports at 3080 [size=32] 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Control ler #2 (rev 02) (prog-if 20 [EHCI]) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, medium devsel, latency 0, IRQ 185 Memory at e0325400 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, fast devsel, latency 0, IRQ 58 Memory at e032 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable - Capabilities: [70] Express Unknown type IRQ 0 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (r ev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: 64bit- Queue=0/0 Enable + Capabilities: [90] #0d [] Capabilities: [a0] Power Management version 2 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (r ev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: 64bit- Queue=0/0 Enable + Capabilities: [90] #0d [] Capabilities: [a0] Power Management version 2 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (r ev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 I/O behind bridge: 2000-2fff Prefetchable memory behind bridge: e040-e040 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: 64bit- Queue=0/0 Enable + Capabilities: [90] #0d [] Capabilities: [a0] Power Management version 2 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controll er #1 (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corporation Unknown device d701 Flags: bus master, medium devsel, latency 0, IRQ 233 I/O ports at 3060 [size=32]
Re: [asterisk-users] Vicidialnow
The astguiclient/VICIDIAL project is very much alive. There were several updates to the 2.0.4 release put out over the last year and we have actually been so busy and so much has changed in the development trunk that we have not had time to put out a full proper release. The latest stable code is of course available through our SVN server as well and most companies that we do installs for are working off of the SVN codebase. Thanks, MATT--- On 1/26/09, David fire ddf...@gmail.com wrote: the vicidial proyect is alive? the last stable releace is from 2007 thanks David 2009/1/26 David fire ddf...@gmail.com where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Vicidialnow
You can download the free manuals and buy the full paid-for manuals at: http://www.eflo.net MATT--- On 1/26/09, David fire ddf...@gmail.com wrote: where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
2009/1/26 Benoit maver...@maverick.eu.org Olivier a écrit : Anyway, according your own experience, how frequent is this PTP case ? I can't say, i don't have any other experience than our own lines That's the point : as much as possible, we're trying to avoid any re-configuration of legacy equipment. So, how would you order TE/NT, PtP/PtMP combination starting from the most useful (for yourself) to the least one. For instance, myself, I would classify them like this : 1. TE PtMP (the most needed) 2. NT PtMP 3. TE PtP 4. NT PtP (the least needed) As for me it is 1. TE PtP 2. NT PtP 3. TE PtMP 4. NT PtMP At least until i encounter the need of a NT/TE PtMP setup. But i don't understand your point, NT/TE PtP and TE PtMP seem to be already implemented, i fail to see the use of classify them, since the only missing is NT PtMP The goal is to provide Matt and Digium with valuable input as thinking a feature is needed is one thing, having several people asking for it is another story ... From Matt's point of view NT PtMP was the least needed ... and so is yours ... Thanks, anyway for telling as at least, it reflects your needs. You want NT PtMP and i second that, not being limited on the asterisk side is a must in the telephony ecosystem, since the legacy PABX aren't alwsys easy to reconfigure. ___ -- 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] German date format in voicemail emails
Did you read the source for app_voicemail? Line 239 says you have to set locale in the config and have the sound file einE. Of course an easier way would be to locate the 19 day and month files and just replace them with German equivalents (assuming that 26 and 2009 sound the same in a German pronunciation). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion Sent: Monday, January 26, 2009 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] German date format in voicemail emails Philipp Kempgen schrieb: Klaus Darilion schrieb: I want to configure voicemail to send emails with the date of the message in German/Austria, that means: Montag, 26 Jänner 2009 instead of Monday, 26 January 2009 voicemail.conf refers to man strftime. This refers to the current locales. So, I tried export LANG=de export LC_ALL=de_DE before starting Asterisk. Unfortunately the date format is still in English. Any hints how to achieve the German language date format? Do you have these locales installed? And what's the default locale on your system? Meanwhile yes :-) Debian: aptitude install locales dpkg-reconfigure locales (Locales to be generated: check the ones you want to install) (Default locale for the system environment: e.g. de_DE.UTF-8) However I'm not sure if that helps. Maybe Asterisk always uses the C locale to expand emaildateformat. I tried setting the locale in app_voicemail manually (ported from minivm) and then it works. But now I try setting it in the environment before starting Asterisk, but that does not work . klaus ___ -- 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] Which policy for ISDN BRI support in NT/PtMP ?
Olivier a écrit : Anyway, according your own experience, how frequent is this PTP case ? I can't say, i don't have any other experience than our own lines That's the point : as much as possible, we're trying to avoid any re-configuration of legacy equipment. So, how would you order TE/NT, PtP/PtMP combination starting from the most useful (for yourself) to the least one. For instance, myself, I would classify them like this : 1. TE PtMP (the most needed) 2. NT PtMP 3. TE PtP 4. NT PtP (the least needed) As for me it is 1. TE PtP 2. NT PtP 3. TE PtMP 4. NT PtMP At least until i encounter the need of a NT/TE PtMP setup. But i don't understand your point, NT/TE PtP and TE PtMP seem to be already implemented, i fail to see the use of classify them, since the only missing is NT PtMP You want NT PtMP and i second that, not being limited on the asterisk side is a must in the telephony ecosystem, since the legacy PABX aren't alwsys easy to reconfigure. ___ -- 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] * Queues with legacy pbx extensions ?
hi we ALWAYS use sip phone IP*-E1-pstn or sip phone-IP-*-E1-legasypbx-?- pstn we use the digiums cards whit echo canceller and we havent any echo problem. more than 20 or 30 installations. whit almost every provider in the country. dont be SO scared. David 2009/1/26 Sriram d_r_sri...@hotmail.com Hello Everybody I am using Trixbox 2.4 (with TE420P PRI lines) .. my setup is like Calls --Asterisk--legacy pbx---analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents like 102,103,104 are present. Now my question is (might be very silly) : 1. I dont have a queue stats software for my legacy pbx, the agents stil logon and logout on their legacy pbx..I was wondering if its possible that all these 102,103,104 etc login to a queue which resides on asterisk ? Can they login and logout of the asterisk queue..I am asking this since there are very nice asterisk queue reporting utilities out there for free..and i wud like to use them ...this will also help me using screen pop-ups for my agents...i know SIP extensions can simplify my setup but echo problems scare me also i dont want to throw my legacy pbx on which i invested heavily.. Can anyone throw some pointers ? Thanks in advance Sriram ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Auto Detect
Just to reply to myself and top post in the process. Its seems this is not the case on CentOS systems my earlier post was the part of the output from a box running Debian Lenny. lspci -v on a CentOS-5.1 box shows neither Kernel driver in use or Kernel modules. Bails bails wrote: Probably not much use to you in this instance, but its always nice to know that lspci -v prints the kernel driver in use and kernel modules. e.g. 10:00.0 Ethernet controller: Broadcom Corporation NetLink BCM5787M Gigabit Ethernet PCI Express (rev 02) Subsystem: Hewlett-Packard Company Device 30c2 Flags: bus master, fast devsel, latency 0, IRQ 220 Memory at d000 (64-bit, non-prefetchable) [size=64K] Expansion ROM at ignored [disabled] Capabilities: access denied Kernel driver in use: tg3 Kernel modules: tg3 Bails David @ULC wrote: lspci ~ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contoller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) but *service network restart RETURNS me* *** Shutting down interface eth0: [ OK ]** Shutting down loopback interface: [ OK ]** Bringing up loopback interface: [ OK ]** Bringing up interface eth0: [ OK ]** Bringing up interface eth1: skge device eth1 does not seem to be present,** delaying initialization.** [FAILED]* On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Suggestion for a new server for E1 line
Hi All, I'm trying to identify a new server as a replacement for what our customer actually has (DELL PowerEdge 860). The server will mount the Digium board TE121, we already have, with echo cancel onboard. I need to know if someone could suggest a new server that's compatible with this board. With compatible I mean that's not having any problem like IRQ sharing, IRQ miss or kernel panic with DAHDI/Zaptel drivers or unwanted hangup or noise during the conversation. My other needs are: 1. At least two RAID disk (preferably hot-swappable but I'm looking also for solution without this feature) 2. Possibly with redundant power supply 3. 1U or 2U size 4. As cheap as possible. Our customer is pushing to have the HP Proliant DL120 but I think it's not fitting the 24/7 needs it has. The server will be used to dispatch calls coming from an 800 free number for one humanitarian organization in Italy. Any suggestion is really welcomed. Thank you very much. Best regards, Marco Signorini http://www.ingegnitech.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] Passing DTMF
I have had this same problem using via:talk. Even though they tell me they have hundreds of people using Asterisk with their service that have no problems we cannot make it work. I have also had reponses confirming that in this email list. So don't wast your time with via:talk. Christopher Gray wrote: Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the question: does anybody know of a carrier that can reliably allow an extension in my pbx to send touchtone to a calling party? I have tried Vitelity and VoicePulse. Neither can do this, and VoicePulse indicates they know it's a problem and will fix it at some unknown time in the future. For the curious, here is the reason for the need. My wife, who works as a translator, will use this extension to receive calls from companies needing translation. When she receives such a call, step 1 for her is to enter an employee id code. At the end of the call, she must enter an additional code to receive an ending time. Vitelity can't do this at all. VoicePulse works about 75% of the time which is not acceptable. Thanks for any advice. Chris Christopher Gray, President Bay Area Digital Promoting good health with innovative technology 870 Market Street, #653 San Francisco, CA 94102 Phone: (415) 217-6667 fax:(415) 962-2520 Email: ch...@bayareadigital.us ___ -- 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 -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- 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] Strange Cisco/Asterisk anomaly
Hey all, having an extremely odd issue wondering if anyone else has come across or seen something similar and what your resolution was. I have Asterisk 1.2.12.1 running (don't ask) on a machine. All has been working fine for months on end. The system has a mixture of Polycom, Snom's and Cisco 7960's running. After a brief power outage last Friday, most phones went down but the PBX stood up (power generator). Anyhow, right now all of the other phones work just fine, but the Cisco's are acting up. My return time is: (numbers sanitized) (sip show peers) 1133/1133 10.10.2.85 D 5060 OK (69 ms) 2217/2217 10.10.2.81 D 5060 OK (15 ms) / 10.10.2.82 D 5060 OK (16 ms) 1137/1137 10.10.2.203 D 5060 OK (202 ms) 3329/3329 10.10.2.51 D 5060 OK (214 ms) 3328/3328 10.10.4.34 D 5060 OK (214 ms) 3345/3345 10.10.2.79 D 5060 OK (186 ms) 3312/3312 10.10.2.127 D 5060 OK (12 ms) 3305/3305 10.10.2.126 D 5060 OK (14 ms) 2269/2269 10.10.2.107 D 5060 OK (34 ms) 2267/2267 10.10.2.134 D 5060 OK (12 ms) 2266/2266 10.10.2.132 D 5060 OK (12 ms) 2207/2207 10.10.2.142 D 5060 OK (16 ms) 3341/3341 10.10.2.118 D 5060 OK (33 ms) 1150/1150 10.10.2.185 D 5060 OK (205 ms) 3304/3304 10.10.2.53 D 5060 OK (142 ms) 3339/3339 10.10.2.139 D 5060 OK (12 ms) 1104/1104 10.10.2.235 D 5060 OK (12 ms) All of the high return time (ms) are Cisco's. When someone dials, they get a fast busy. Asterisk doesn't even see them registering some times. They're flapping up and down sometimes @ 712ms. So to be fair, switches and routers were rebooted. However, all of these phones are on the same switch so for example ext 3328 and 1104 are on the same switch, same VLAN, no ACL's, no fw rules, nada. The only differentiator between these phones are the high MS times are Cisco phones. Any thoughts? Possible firmware corruption? I'm at a loss to figure this one out. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP Enough research will tend to support your conclusions. - Arthur Bloch A conclusion is the place where you got tired of thinking - Arthur Bloch 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E ___ -- 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] Strange Cisco/Asterisk anomaly
You've tried a sip reload from CLI and rebooted the phones? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo Sent: Monday, January 26, 2009 8:51 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange Cisco/Asterisk anomaly Hey all, having an extremely odd issue wondering if anyone else has come across or seen something similar and what your resolution was. I have Asterisk 1.2.12.1 running (don't ask) on a machine. All has been working fine for months on end. The system has a mixture of Polycom, Snom's and Cisco 7960's running. After a brief power outage last Friday, most phones went down but the PBX stood up (power generator). Anyhow, right now all of the other phones work just fine, but the Cisco's are acting up. My return time is: (numbers sanitized) (sip show peers) 1133/1133 10.10.2.85 D 5060 OK (69 ms) 2217/2217 10.10.2.81 D 5060 OK (15 ms) / 10.10.2.82 D 5060 OK (16 ms) 1137/1137 10.10.2.203 D 5060 OK (202 ms) 3329/3329 10.10.2.51 D 5060 OK (214 ms) 3328/3328 10.10.4.34 D 5060 OK (214 ms) 3345/3345 10.10.2.79 D 5060 OK (186 ms) 3312/3312 10.10.2.127 D 5060 OK (12 ms) 3305/3305 10.10.2.126 D 5060 OK (14 ms) 2269/2269 10.10.2.107 D 5060 OK (34 ms) 2267/2267 10.10.2.134 D 5060 OK (12 ms) 2266/2266 10.10.2.132 D 5060 OK (12 ms) 2207/2207 10.10.2.142 D 5060 OK (16 ms) 3341/3341 10.10.2.118 D 5060 OK (33 ms) 1150/1150 10.10.2.185 D 5060 OK (205 ms) 3304/3304 10.10.2.53 D 5060 OK (142 ms) 3339/3339 10.10.2.139 D 5060 OK (12 ms) 1104/1104 10.10.2.235 D 5060 OK (12 ms) All of the high return time (ms) are Cisco's. When someone dials, they get a fast busy. Asterisk doesn't even see them registering some times. They're flapping up and down sometimes @ 712ms. So to be fair, switches and routers were rebooted. However, all of these phones are on the same switch so for example ext 3328 and 1104 are on the same switch, same VLAN, no ACL's, no fw rules, nada. The only differentiator between these phones are the high MS times are Cisco phones. Any thoughts? Possible firmware corruption? I'm at a loss to figure this one out. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP Enough research will tend to support your conclusions. - Arthur Bloch A conclusion is the place where you got tired of thinking - Arthur Bloch 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E ___ -- 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] Suggestion for a new server for E1 line
On Mon, 26 Jan 2009, Marco Signorini wrote: Hi All, I'm trying to identify a new server as a replacement for what our customer actually has (DELL PowerEdge 860). The server will mount the Digium board TE121, we already have, with echo cancel onboard. I need to know if someone could suggest a new server that's compatible with this board. With compatible I mean that's not having any problem like IRQ sharing, IRQ miss or kernel panic with DAHDI/Zaptel drivers or unwanted hangup or noise during the conversation. My other needs are: 1. At least two RAID disk (preferably hot-swappable but I'm looking also for solution without this feature) 2. Possibly with redundant power supply 3. 1U or 2U size 4. As cheap as possible. Our customer is pushing to have the HP Proliant DL120 but I think it's not fitting the 24/7 needs it has. The server will be used to dispatch calls coming from an 800 free number for one humanitarian organization in Italy. Cheap and Redundant Power Supply don't go well together... It might sometimes be cheaper to build 2 units than get a redundant PSU - not the same type of redundancy, however... And you can boot/run from SSD (Solid State Drive) - You should easilly fit a standard system into a 16 or 32GB Flash module these days. Something like these if you can find an italian supplier: http://linitx.com/viewcategory.php?catid=1011pp=100,1011 Not as cheap as 2 drives though. Gordon ___ -- 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] Strange Cisco/Asterisk anomaly
On Mon, 26 Jan 2009, Danny Nicholas wrote: You've tried a sip reload from CLI and rebooted the phones? Been there, done that. Rebooted the phones manually, switches manually, wiresharked for anomalies, nothing out of the ordinary. Since as stated there was an outage, the server maintained power. I decided to reboot the server itself 10 minutes ago and all was resolved - so that's something odd in of itself. I configured Asterisk in a high availability cluster so there is always a failover however, in this case, just one of the things I like to view (well my shop likes to also) as - an Asterisk fluke. I'd like to upgrade the system, but it would need to be a complete revamp as bringing it up to par would mean redoing Zaptel, etc. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP Enough research will tend to support your conclusions. - Arthur Bloch A conclusion is the place where you got tired of thinking - Arthur Bloch 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E ___ -- 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] Strange Cisco/Asterisk anomaly
Another possible workaround/hack (since this is an older *) would be to do a cron asterisk -rx restart when convenient for around midnight local time each day just to clear the cobwebs. Unless something was really hosed, this would clean up things faster and easier than a server restart. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo Sent: Monday, January 26, 2009 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Strange Cisco/Asterisk anomaly On Mon, 26 Jan 2009, Danny Nicholas wrote: You've tried a sip reload from CLI and rebooted the phones? Been there, done that. Rebooted the phones manually, switches manually, wiresharked for anomalies, nothing out of the ordinary. Since as stated there was an outage, the server maintained power. I decided to reboot the server itself 10 minutes ago and all was resolved - so that's something odd in of itself. I configured Asterisk in a high availability cluster so there is always a failover however, in this case, just one of the things I like to view (well my shop likes to also) as - an Asterisk fluke. I'd like to upgrade the system, but it would need to be a complete revamp as bringing it up to par would mean redoing Zaptel, etc. =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP Enough research will tend to support your conclusions. - Arthur Bloch A conclusion is the place where you got tired of thinking - Arthur Bloch 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E ___ -- 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] Ntework Card
This is still very off topic... Someone's already suggested you look to somewhere for centos help... if you had, you'd have found this... http://www.centos.org/modules/newbb/viewtopic.php?forum=39topic_id=10098 Which is an RPM containing an update driver, so you wouldn't have to mess about poluting your system... Also, you'd have found documentation which would tell you how to setup a NIC manually... I'd guess you missed the bit about editing modprobe.conf... If you ask these question in the right place, you'll probably get much better help... d 2009/1/26 David @ULC ucoms2...@gmail.com [r...@vicidialnow ~]# cat /etc/sysconfig/network-scripts/ifcfg-eth1 NAME= BOOTPROTO=none TYPE=Ethernet DEVICE=eth1 MTU= NETMASK=255.255.255.0 BROADCAST=192.168.0.100 IPADDR=192.168.0.100 NETWORK=192.168.0.0 ONBOOT=no ___ -- 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] Auto Detect
on CentOS you can use lshal to return which driver is in use... e.g. lshal -s |grep pci_ |xargs -n1 lshal -l -u |grep -E udi|info.product|info.linux.driver get's a list of items, filters on pci, gets the long output for those pci devices and outputs lines containing udi, info.product and info.linux.driver... results returned like the following (but with more devices obviously)0... udi = '/org/freedesktop/Hal/devices/pci_10b7_9200' info.udi = '/org/freedesktop/Hal/devices/pci_10b7_9200' (string) info.product = '3c905C-TX/TX-M [Tornado]' (string) info.linux.driver = '3c59x' (string) udi = '/org/freedesktop/Hal/devices/pci_10ec_8139' info.udi = '/org/freedesktop/Hal/devices/pci_10ec_8139' (string) info.product = 'RTL-8139/8139C/8139C+' (string) info.linux.driver = '8139too' (string) d 2009/1/26 bails ba...@westcomuk.com Just to reply to myself and top post in the process. Its seems this is not the case on CentOS systems my earlier post was the part of the output from a box running Debian Lenny. lspci -v on a CentOS-5.1 box shows neither Kernel driver in use or Kernel modules. Bails bails wrote: Probably not much use to you in this instance, but its always nice to know that lspci -v prints the kernel driver in use and kernel modules. e.g. 10:00.0 Ethernet controller: Broadcom Corporation NetLink BCM5787M Gigabit Ethernet PCI Express (rev 02) Subsystem: Hewlett-Packard Company Device 30c2 Flags: bus master, fast devsel, latency 0, IRQ 220 Memory at d000 (64-bit, non-prefetchable) [size=64K] Expansion ROM at ignored [disabled] Capabilities: access denied Kernel driver in use: tg3 Kernel modules: tg3 Bails David @ULC wrote: lspci ~ 00:00.0 Host bridge: Intel Corporation 82G35 Express DRAM Controller (rev 03) 00:02.0 VGA compatible controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation 82G35 Express Integrated Graphics Controller (rev 03) 00:19.0 Ethernet controller: Intel Corporation 82566DC Gigabit Network Connection (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Contoller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 IDE interface: JMicron Technologies, Inc. JMB368 IDE controller 04:03.0 Ethernet controller: D-Link System Inc DGE-530T Gigabit Ethernet Adapter (rev 11) (rev 11) 04:05.0 FireWire (IEEE 1394): Agere Systems FW323 (rev 70) but *service network restart RETURNS me* *** Shutting down interface eth0: [ OK ]** Shutting down loopback interface: [ OK ]** Bringing up loopback interface: [ OK ]** Bringing up interface eth0: [ OK ]** Bringing up interface eth1: skge device eth1 does not seem to be present,** delaying initialization.** [FAILED]* On Mon, Jan 26, 2009 at 5:24 PM, David @ULC ucoms2...@gmail.com wrote: Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk ___ -- 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.
[asterisk-users] goto iax problem
Dear, the goto function to the iax dialing, makes bill duration and call duration wrong, in cdr.they are equal to ringing time. the cdr will be produced and saved into the dbase, when the callee picks up the phone. is any way to have real duration time ? [main] exten = _1X.,1,GOTO(LOPL,${EXTEN},1) [LOPL] exten = _X.,1,Dial(IAX2/MAIN/${EXTEN},60) ___ -- 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] CDR 0.00 duration
On Wed, 2009-01-21 at 15:45 +0530, Sriram wrote: Hi I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls that have duration of 0 seconds, but they are still shown as ANSWERED . how come its possible when duration is 0.00 ? Are the callers billed for such calls ? Rgds Sriram Sriram-- Well, if the end or ANSWER time isn't set, then you would get a 0 duration. murf -- Steve Murphy m...@digium.com Digium smime.p7s Description: S/MIME cryptographic 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] Network Card
Here is the content of modprobe.conf : alias eth1 skge alias scsi_hostadapter ata_piix alias snd-card-0 snd-hda-intel options snd-card-0 index=0 options snd-hda-intel index=0 remove snd-hda-intel { /usr/sbin/alsactl store 0 /dev/null 21 || : ; }; /sbin/modprobe -r --ignore-remove snd-hda-intel ___ -- 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] custom cdr userfiled
Pezhman Lali pezhman_l...@yahoo.com writes: so great, I am using odbc . but I can not find custom_odbc . is any keywords or links to useful documents? Top posting is annoying. cdr_adaptive_odbc is what you want. It's available in 1.6.x, and there's a backport for 1.4.x. I haven't tried the backport, but it works very nicely in 1.6.x. /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
[asterisk-users] General Asterisk SIP/IAX provider question
My coworker and I have built an Asterisk box. Everything went well, now we are ready to hook the box to a SIP/IAX provider. Does anyone have recommendation on choosing a vendor? We are located in Virginia. Thanks Shane ___ -- 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] General Asterisk SIP/IAX provider question
Shane, You can try, www.didforsale.com. We allow free testing with no purchase required. See what others are saying, http://www.didforsale.com/blog/?p=103 -Jai On Mon, Jan 26, 2009 at 11:12 AM, Thomas Mullins tsmull...@wise.k12.va.uswrote: My coworker and I have built an Asterisk box. Everything went well, now we are ready to hook the box to a SIP/IAX provider. Does anyone have recommendation on choosing a vendor? We are located in Virginia. Thanks Shane ___ -- 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] General Asterisk SIP/IAX provider question
Most thanks, I am there now. Shane From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jai Rangi Sent: Monday, January 26, 2009 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] General Asterisk SIP/IAX provider question Shane, You can try, www.didforsale.com. We allow free testing with no purchase required. See what others are saying, http://www.didforsale.com/blog/?p=103 -Jai On Mon, Jan 26, 2009 at 11:12 AM, Thomas Mullins tsmull...@wise.k12.va.us wrote: My coworker and I have built an Asterisk box. Everything went well, now we are ready to hook the box to a SIP/IAX provider. Does anyone have recommendation on choosing a vendor? We are located in Virginia. Thanks Shane ___ -- 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] Dahdi caused Kernel to segfault
Hi the same happened here also with different distros (ubuntu and fedora 9) each time i run dahdi start the kernel crash. i was using the dahdi from trunk regards, David fire a écrit : do you have any dahdi card ??? if not edit /etc/dahdi/modules so it dosent load any modules. David 2009/1/25 broadband Voice broadbandvo...@gmail.com mailto:broadbandvo...@gmail.com More information service dahdi start Loading DAHDI hardware modules: FATAL: Module dahdi not found. wct4xxp: [ OK ] Error: missing /dev/dahdi! [r...@newmark1 ~]# On Sun, Jan 25, 2009 at 3:31 PM, broadband Voice broadbandvo...@gmail.com mailto:broadbandvo...@gmail.com wrote: I had several panic attacks after upgrading to 1.4.22 but now we have no dial tone on the T1. Urgent production system. On Tue, Jan 13, 2009 at 5:13 AM, Benoit maver...@maverick.eu.org mailto:maver...@maverick.eu.org wrote: Personnaly, i had recently encountered a global machine check exception with two cards (TE220p and B410) and many kernel panic with mISDN (mostly if i tried to unload it). Dahdi still hasn't failed me (directly) Thomas Kenyon a écrit : Yesterday, a low-duty production server that I maintain core-dumped. At the time there were only around 2 calls going through it. The strace on the screen made it look like it was caused by Dahdi. The machine is running asterisk-1.6.0.3 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-addons-1.6.0 Kernel version 2.6.28 There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards. Has anyone had a similar issue? This has only happened once, but I am a bit worried. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Dahdi caused Kernel to segfault
On Mon, Jan 26, 2009 at 08:53:11PM +, Rony Ron wrote: Hi the same happened here also with different distros (ubuntu and fedora 9) each time i run dahdi start the kernel crash. What exactly do you mean by crash? An error, or the system completely crashes / hangs? i was using the dahdi from trunk What revision exactly? try 'svn info' at the source directory. -- 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] Dahdi Init script for Suse?
Nah, not using RPMs. This is a from-source build. Part of the problem is, I'm running Novell's Open Enterprise Server, which is SLES 10.1 (I think) with Novell's OES2 Overlay on top. So the Asterisk RPMs available for that revision of SLES weren't usable for me (they had 1.2 available, and I wanted 1.4), hence, source build. Might play with 1.6 when I start feeling adventurous again, as my asterisk config is really, really simple (it's a bridge between an old Rolm CBX and HylaFax). But I figured, if their init script supports two of the top distros, why not a third? Wasn't sure, so I was seeking comments on the script I attached to my last e-mail. --J -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Saturday, January 24, 2009 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Init script for Suse? On Sat, 2009-01-24 at 23:45 +0100, Marco wrote: Hi, I've it up and running on OpenSuse 11. I used the scripts provided by the sources and commented out one line: # # Determine which kind of configuration we're using # #system=redhat # assume redhat system=debian # assume debian This forces the script to use debian style. It works for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. Just wondering... The O.P. said he's using SLE. You're talking about openSUSE. Are you using the rpm's from the OBS? The zaptel-rpm for 10.3 were containing the proper startup scripts. I've got some suse machines running asterisk, but as soon as hw get's involved, i'm stuck: neither pri, nor bri (mISDN) seems to be working on anything later than 10.3. Hans ___ -- 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
[asterisk-users] Dial weirdness
I'm seeing this response to SIP calls originated in the following manner: Dial(SIP/${EXTEN}SIP/{$DID},30,r) handle_response_invite: Re-invite to non-existing call leg on other UA. The response is from the second part of the dial. What exactly does it mean and how can I fix it? Thanks in advance Bruce ___ -- 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 Init script for Suse?
On Sat, Jan 24, 2009 at 04:08:53PM -0500, Joshua Kinard wrote: Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box that'll work right? The one included by default only deals with debian and redhat, and the changes between the old zaptel script I have that works are far too invasive. Notably in the use of this action command that's probably redhat specific. Anything equivalent in SuSE? The Debian equivalent of action Starting FooBar foobar , if you source /etc/lsb/init-functions is log_daemon_msg Starting FooBar foobar log_end_msg $? Though it has an atvantage of making it easier to redirect output of the command. (Those functions don't seem to be part of the standard LSB init functions, sadly) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial weirdness
Bruce Ferrell wrote: I'm seeing this response to SIP calls originated in the following manner: Dial(SIP/${EXTEN}SIP/{$DID},30,r) handle_response_invite: Re-invite to non-existing call leg on other UA. The response is from the second part of the dial. What exactly does it mean and how can I fix it? Thanks in advance Bruce First of all, it may just be a transcription error on your part, but the variable in the second part of the Dial statement should be ${DID} instead of {$DID}. That message you see on the console probably means that Asterisk has received a 481 response to the INVITE it has sent out. Apparently, whoever is receiving the INVITE thinks that it is a re-INVITE that belongs to an established SIP dialog, but then it can't actually find the dialog to which the INVITE belongs. This seems like it is an incorrect interpretation by the remote end since Asterisk generates a new callid, new from-tag, and has no to-tag on each initial INVITE it sends out when starting a call. It may be helpful to look at a packet capture from a failed attempt. It may be that whoever is sending back the 481 is sending a reason for it, or it may be that there is something obviously malformed in the SIP requests being sent by Asterisk. Mark Michelson ___ -- 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] I need help
i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Document with differences between 1.2, 1.4 and 1.6?
Is there a bullet type document with the features each version of Asterisk has? I know you can read the CHANGES file but that is not something you give a customer. I just need a one or two page document with bullet points showing the features added from 1.2 to 1.4 and from 1.4 to 1.6. Anyone know of an existing document or it this a make your own moment? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] General Asterisk SIP/IAX provider question
Doesn't this belong on the biz list?? Sure hope their service is better than their web site Peg Leg O'Brien Thomas Mullins wrote: Most thanks, I am there now. Shane *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jai Rangi *Sent:* Monday, January 26, 2009 2:25 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] General Asterisk SIP/IAX provider question Shane, You can try, www.didforsale.com http://www.didforsale.com. We allow free testing with no purchase required. See what others are saying, http://www.didforsale.com/blog/?p=103 -Jai On Mon, Jan 26, 2009 at 11:12 AM, Thomas Mullins tsmull...@wise.k12.va.us mailto:tsmull...@wise.k12.va.us wrote: My coworker and I have built an Asterisk box. Everything went well, now we are ready to hook the box to a SIP/IAX provider. Does anyone have recommendation on choosing a vendor? We are located in Virginia. Thanks Shane ___ -- 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 -- Dog is my co-pilot ___ -- 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] Document with differences between 1.2, 1.4 and 1.6?
On 16:28, Mon 26 Jan 09, Carlos Chavez wrote: Is there a bullet type document with the features each version of Asterisk has? I know you can read the CHANGES file but that is not something you give a customer. I just need a one or two page document with bullet points showing the features added from 1.2 to 1.4 and from 1.4 to 1.6. Anyone know of an existing document or it this a make your own moment? Because it depends on what your customers use and what your customers need it's basically a 'make your own' task. Reading the UPGRADE-1.X.txt files gives you a nice overview of what you have to do differently and what happened between major releases. -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? pgpFzZXvpYZ5b.pgp 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] I need help
And your problem is... ? Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] I need help
and what is your problem? if your email is shorter than your signature maybe you forgot something. David 2009/1/26 Bayardo Sanchez bayardo.sanc...@gmail.com i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] I need help
There are no good Mexican restaurants near my house. PaulH Jose P. Espinal wrote: And your problem is... ? Bayardo Sanchez wrote: i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 ___ -- 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] I need help
Bayardo you need to ask questions but exact, nobody has a glass ball to help you regardss rickygm 2009/1/26 Bayardo Sanchez bayardo.sanc...@gmail.com i have a problem need help == Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on 'SIP/8022-b7225740' -- Got SIP response 503 Service Unavailable back from 74.63.41.218 -- SIP/voipms4-09ab0c38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/8011-b724f888' status is 'CONGESTION' == Spawn extension (DLPN_everything, 8312549244, 2) exited non-zero on 'SIP/8010-b72241b0' -- rickygm ___ -- 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] German date format in voicemail emails
On Monday 26 January 2009 08:21:10 am Danny Nicholas wrote: Did you read the source for app_voicemail? Line 239 says you have to set locale in the config and have the sound file einE. Of course an easier way would be to locate the 19 day and month files and just replace them with German equivalents (assuming that 26 and 2009 sound the same in a German pronunciation). You might want to read his message before you start recommending things. The OP was interested in emails, not message playback. The language setting only affects prompt playback, not email messages. -- Tilghman ___ -- 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] Network Card
Solved. Here is the Solution : http://www.centos.org/modules/newbb/viewtopic.php?topic_id=18299start=0#forumpost66716 On Mon, Jan 26, 2009 at 6:23 PM, David @ULC ucoms2...@gmail.com wrote: Sorry, but any further help withhttp://lists.digium.com/pipermail/asterisk-users/2009-January/225548.html http://lists.digium.com/pipermail/asterisk-users/2009-January/225548.html ___ -- 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] Reading/Writing the Astdb
Tzafrir; That's a really good idea, however, I am having problems getting it to work. I tried the following: echo -n asterisk -rx \database put FOO BAR 1\ | socat - /var/run/asterisk/asterisk.ctl and echo -n asterisk -rx \database put FOO BAR 1\ | socat - UNIX-CONNECT: /var/run/asterisk/asterisk.ctl both look like there was a connection to asterisk on the CLI, but no update. What am I missing? BTW; the difference between connecting to asterisk by doing a asterisk -rx cmd and doing it this way was 500ms the first way and 7ms the second. Awesome. Date: Sat, 24 Jan 2009 23:23:40 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reading/Writing the Astdb On Sat, Jan 24, 2009 at 11:00:58AM -0500, cbbs...@hotmail.com wrote: All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. Either do that through the manager interface, or (if you want to batch commands) send them directly over the unix-domain socket asterisk.ctl . -- 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 _ Windows Live™: E-mail. Chat. Share. Get more ways to connect. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_allup_howitworks_012009___ -- 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] Reading/Writing the Astdb
At 8:48 PM on 26 Jan 2009, cbbs...@hotmail.com wrote: Tzafrir; That's a really good idea, however, I am having problems getting it to work. I tried the following: echo -n asterisk -rx \database put FOO BAR 1\ | socat - /var/run/asterisk/asterisk.ctl and echo -n asterisk -rx \database put FOO BAR 1\ | socat - UNIX-CONNECT: /var/run/asterisk/asterisk.ctl both look like there was a connection to asterisk on the CLI, but no update. What am I missing? BTW; the difference between connecting to asterisk by doing a asterisk -rx cmd and doing it this way was 500ms the first way and 7ms the second. Awesome. Just guessing, but try it without the asterisk -rx: echo -n database put FOO BAR 1 | socat - UNIX-CONNECT:/var/run/asterisk/asterisk.ctl And if that doesn't work, maybe omit the -n so that it sends a newline. Date: Sat, 24 Jan 2009 23:23:40 +0200 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reading/Writing the Astdb On Sat, Jan 24, 2009 at 11:00:58AM -0500, cbbs...@hotmail.com wrote: All; I have a question regarding the Astdb. When reading more than a few values, it can take quite a while to grab several values in the astdb using say, asterisk -rx database show output.txt and work with that and then set a new value such as asterisk -rx database put $key $value. The whole process can take over 1 second for EACH ENTRY which adds up for more than a few keys. Either do that through the manager interface, or (if you want to batch commands) send them directly over the unix-domain socket asterisk.ctl . -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.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] Reading/Writing the Astdb
On Mon, 26 Jan 2009, cbbs...@hotmail.com wrote: That's a really good idea, however, I am having problems getting it to work. I tried the following: echo -n asterisk -rx \database put FOO BAR 1\ | socat - /var/run/asterisk/asterisk.ctl and echo -n asterisk -rx \database put FOO BAR 1\ | socat - UNIX-CONNECT: /var/run/asterisk/asterisk.ctl echo -n show channels\ | sudo socat STDIO UNIX-CONNECT:/var/run/asterisk.ctl and echo -n show channels\ | sudo socat STDIO /var/run/asterisk.ctl work for me. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Help with cdr_odbc
I have having a hard time setting the cdr with cdr_odbc. Below is all my conf file related to it, let me know what I am doing wrong. Thank you. *cdr.conf* [general] enable=yes [csv] usegmtime=yes; log date/time in GMT. Default is no loguniqueid=yes ; log uniqueid. Default is no loguserfield=yes ; log user field. Default is no *cdr_odbc.conf** *[global] dsn=MySQL-asterisk loguniqueid=yes dispositionstring=yes table=cdr usegmtime=no username=myuser password=mypass *modules.conf* [modules] autoload=yes preload = res_odbc.so preload = res_config_odbc.so preload = func_strings.so noload = pbx_gtkconsole.so load = res_musiconhold.so load = cdr_odbc.so noload = chan_alsa.so *res_odbc.conf *[ENV] [asterisk] enabled = yes dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes* odbc.ini* [MySQL-asterisk] Description = Asterisk MySQL ODBC Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = myuser Password= mypass Database= asterisk Option = 3 #Port = *odbcinst.ini *[MySQL] Description = MySQL driver Driver = /usr/lib/odbc/libmyodbc.so Setup = /usr/lib/odbc/libodbcmyS.so FileUsage = 1 I have my table cdr in the asterisk database, I am using mysql and asterisk 1.6.0.3. The mysql user has all priviledges granted on host %(any) Any help would be appreciated ___ -- 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] Reading/Writing the Astdb
On Mon, Jan 26, 2009 at 06:31:30PM -0800, Steve Edwards wrote: On Mon, 26 Jan 2009, cbbs...@hotmail.com wrote: That's a really good idea, however, I am having problems getting it to work. I tried the following: echo -n asterisk -rx \database put FOO BAR 1\ | socat - /var/run/asterisk/asterisk.ctl and echo -n asterisk -rx \database put FOO BAR 1\ | socat - UNIX-CONNECT: /var/run/asterisk/asterisk.ctl echo -n show channels\ | sudo socat STDIO UNIX-CONNECT:/var/run/asterisk.ctl and echo -n show channels\ | sudo socat STDIO /var/run/asterisk.ctl For a loop, though, you'll have to make sure each command is sent in its own separate call to write() . The way I used was to put a 'sleep 0.001' after each command. Also note that you don't get the output of the command, nor is there any error handling. -- 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
[asterisk-users] hangup problem(for spa400)
Hi all, I have asterisk connected to my voice application server. Asterisk is connected and registering to a linksys spa400 box. I am running an application on a perticular extention (141). Here is a snip from my extensions.conf... exten = spa400,s,MyApp(/etc/asterisk/MyAppConfig.conf) exten = spa400,s+1,Hangup when an incoming call comes,It is accepted properly,And the application executes successfully,Also I see SIP BYE going to spa400 and getting a 200 Ok for BYE from spa400.But the problem is that the call is not really disconnected,Caller's billing doesn't stop, It takes atleast a minute,before the 'call end tone' can be heard by the caller. It can be a spa400 issue,(As I can see a SIP BYE at asterisk end). Seeking for some help/pointers Thanks -- === (-: Saurabh :-) === French is the language of love,For everything else there is 'C' Every search begins with beginner's luck and ends with the victor being severly tested -Paulo Coehlo ___ -- 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] Can't start Asterisk after installing Digium G729 licence
Hi, I carefully followed instructions in README file lasting with : /root/register ... blabla asterisk -r CLI restart now Then asterisk -r fails with : # asterisk -r Asterisk 1.6.1-beta4, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Running as group 'asterisk' == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf == Found Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) # tail /var/log/asterisk/messages [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasiax at line 39. [Jan 27 08:17:40] WARNING[23451] chan_dahdi.c: Ignoring hasmanager at line 47. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: G.729 transcoding module version 34, Copyright (C) 1999-2007 Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Please see the full license text supplied by the accompanying [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: register utility, or ask for a copy from Digium. [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: This product includes software developed by the OpenSSL Project [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jan 27 08:17:40] NOTICE[23451] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project Before opening a ticket to Digium, is there something obvious I missed ? Google didn't show much hint ... After reading this see register utility line, I ran /root/register once again but it didn't change anything ... Cheers ___ -- 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