Re: [asterisk-users] chan_capi install problems
On Sun, 27 May 2007, CSB wrote: The current RPM ftp://ftp.melware.net/Trixbox/chan_capi-1.0.1_1.2.14-1.i386.rpm installs but This precompiled RPM is for the previous trixbox asterisk version 1.2.14. A new RPM will follow soon... I look forward to it. If you want to compile chan-capi by yourself, you need to install all dev- packages to have the needed header files. I think this should do it: yum -y install isdn4k-utils-devel asterisk-devel Having done that, I now get a message on asterisk startup: May 27 21:23:43 VERBOSE[4288] logger.c: [chan_capi.so]May 27 21:23:43 WARNING[4288] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_pickup_call May 27 21:23:43 WARNING[4288] loader.c: Loading module chan_capi.so failed! new chan-capi uses ast_pickup_call too. But this function is provided by module res_features. So you need to make sure to load res_features before chan-capi is loaded, e.g. in modules.conf: [modules] load=res_features.so load=chan_capi.so Armin But if the trixbox asterisk version again has special patches applied (something like jitterbuffer patch) which is not known to external modules like chan-capi, the compiled chan-capi may cause craches because it just doesn't match with the configured asterisk header files. I am intending to use Trixbox but in the meantime for testing purposes have installed Asterisk from source. Any further advice appreciated. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet). It worked fine for the testing I was doing. I'm not sure of the status or performance of the PCI mapping through to DomU these days, but that should be the only extra step required. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roberto Pereyra Sent: Saturday, 26 May 2007 23:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP accounts from MYSQL.
Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP accounts from MYSQL.
Asterisk realtime is what you are looking for. the subject is explained very clearly including configuration examples and DB schema on the following links: http://www.voip-info.org/wiki-Asterisk+RealTime http://www.asteriskdocs.org/modules/news/article.php?storyid=28 I won't go over the process as it is detailed in the links above, but basically you should compile the asterisk-addons, configure the res_mysql with the proper DB details, create a table to hold sip.conf and optionally extensions.conf then configure extconfig to map the newly created tables. Joss. On 5/27/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI problem, pri_fixup_principle: Call specified, but not found? [SOLVED]
Issue was that zapata.conf was specifying trunkgroup for the spans. This was adding a DS1 Identifier to the setup channel ID, which the CO does not recognize and so it assigns its own pick of channel. This was causing a move which asterisk does not cope with. (Actually, initial channel id being exclusive explicitly forbids B channel being moved). As Matt said, configuration error. -Carlos Carlos G Mendioroz @ 24/05/2007 10:33 -0300 dixit: Here... Please advise if any special flags/options are needed. [ 02 01 01 27 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 019 P/F: 1 0 bytes of data -- ACKing all packets from 18 to (but not including) 19 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (51) [ 02 01 01 67 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 051 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (51) [ 00 01 01 67 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 051 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 27 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 019 P/F: 1 0 bytes of data -- ACKing all packets from 18 to (but not including) 19 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter [ 02 01 01 b1 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 088 P/F: 1 0 bytes of data -- ACKing all packets from 87 to (but not including) 88 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (6) [ 02 01 01 0d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 006 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (6) [ 00 01 01 0d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 006 P/F: 1 0 bytes of data -- Restarting T203 counter [ 00 01 01 b1 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 088 P/F: 1 0 bytes of data -- ACKing all packets from 87 to (but not including) 88 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter -- Accepting AUTHENTICATED call from 10.8.0.6: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (), priority = mine -- Executing Dial(IAX2/10.8.0.6:4569-3, Zap/g2/113) in new stack -- Requested transfer capability: 0x00 - SPEECH [ 00 01 26 66 08 02 0e 89 05 04 03 80 90 a3 18 04 e9 82 83 81 28 08 43 61 72 6c 6f 73 20 4d 6c 06 41 81 31 31 30 30 70 04 c1 31 31 33 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 019 0: 0 N(R): 051 P: 0 41 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 3721/0xE89) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 04 e9 82 83 81] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan:0 ChanSel: Reserved Ext: 1 DS1 Identifier: 2 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 08 43 61 72 6c 6f 73 20 4d] Display (len= 8) [ Carlos M ] [6c 06 41 81 31 31 30 30] Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening
Re: [asterisk-users] reset Polycom phones remotely
Forum wrote on 5/26/07 5:32 PM: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Have you checked their boot server type, and does it match what you have available? If FTP is all you have set up on the boot server and those two phones are set to use TFTP then you would have this issue. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP accounts from MYSQL.
Than you Joss, the links was very usefull. On 5/27/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: Asterisk realtime is what you are looking for. the subject is explained very clearly including configuration examples and DB schema on the following links: http://www.voip-info.org/wiki-Asterisk+RealTime http://www.asteriskdocs.org/modules/news/article.php?storyid=28 I won't go over the process as it is detailed in the links above, but basically you should compile the asterisk-addons, configure the res_mysql with the proper DB details, create a table to hold sip.conf and optionally extensions.conf then configure extconfig to map the newly created tables. Joss. On 5/27/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.18 problem
hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type asterisk -r but the response is Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?). how to solve this. thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Xen domu with tdm400 hardware
Why would you want to do this? If you wanted to run multiple systems together on an Asterisk server I would run the Asterisk server on Dom0 and the other stuff on DomU systems. -Jonathan James Harper wrote: I did it back in the xen 2.x days with a BRI adapter (Traverse NetJet). It worked fine for the testing I was doing. I'm not sure of the status or performance of the PCI mapping through to DomU these days, but that should be the only extra step required. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roberto Pereyra Sent: Saturday, 26 May 2007 23:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk in Xen domu with tdm400 hardware Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zonbu
I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP accounts from MYSQL.
I don't think you need to modify th Makefile at all. That might be why you are having trouble finding details on that. -Jonathan Jonson Player wrote: Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql. Thank you for your support guys. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] reset Polycom phones remotely
It's definitely ftp. I have given the phone a static ip. When I set it to dhcp it just hangs and cannot get an IP. I can ping the phone and see the web config page so it is on the network. Any more suggestions. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller Sent: Sunday, 27 May 2007 5:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reset Polycom phones remotely Forum wrote on 5/26/07 5:32 PM: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Have you checked their boot server type, and does it match what you have available? If FTP is all you have set up on the boot server and those two phones are set to use TFTP then you would have this issue. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.18 problem
On Sun, May 27, 2007 at 05:43:59PM +0200, MOSBAH ABDELKADER wrote: hello, I have installed asterisk 1.2.18 in suse 10.2. After typing asterisk in the terminal command line (i don't think that asterisk runs when doing this) i type asterisk -r but the response is Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?). Is asterisk running? If it is not running (i.e, configuration file missing somewhere) then you need to correct that. Check the permissions on the file /var/run/asterisk.ctl. If you are running asterisk -r as a non-root user, then you need to make sure that user has permission (group etc.) to read/write this fiel. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h323friends peer realtime
Can anyone help me to make h323friends peer realtime ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.18 problem
asterisk (does /var/run/asterisk.ctl exist?). This connects to an already running asterisk system. You first have to start asterisk, type this: asterisk -vc Or I put the script in /etc/init.d and tell the system to start it on boot. Then you can do asterisk -r ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zonbu
On Sun, May 27, 2007 at 11:52:32AM -0400, Dean Collins wrote: I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html They don't seem to encourge you to replace their OS with one of your own. There is no real hardware specification. The CPU i probably some VIA but might as well be something else. Probably not good for too many calls. I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. The price is not exactly 100$. The minimal price seems to be 250$, as it includes a monthly subscription of 12.95$ for a period of 2 years. At that price range there are some other nice systems. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Divitas
I was cleaning through some old IT magazines this long weekend when I came across a company called Divitas in the April 30th edition of Network Computing. I've never heard of them but has anyone else heard of them? Basically they have a call control appliance that can deliver centrally held up calls between not only GSM but also redirect the call to a wifi hotspot if you are in range. It seems like a neat concept that shouldn't necessarily be beyond the capabilities of Asterisk (apart from the fact that the end Win Mobile 5 / Symbian handset would need some type of client). Any thoughts? At $550 per seat looks an expensive way to transfer calls between networks but I've never seen another CPE piece of equipment that can do this. http://www.divitas.com/products Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation image001.gif___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Divitas
There will be a number of companies set to offer similar services. In 3 months we will have a 24 port SIP-GSM-SKYPE gateway -E On 5/27/07, Dean Collins [EMAIL PROTECTED] wrote: I was cleaning through some old IT magazines this long weekend when I came across a company called Divitas in the April 30th edition of Network Computing. I've never heard of them but has anyone else heard of them? Basically they have a call control appliance that can deliver centrally held up calls between not only GSM but also redirect the call to a wifi hotspot if you are in range. It seems like a neat concept that shouldn't necessarily be beyond the capabilities of Asterisk (apart from the fact that the end Win Mobile 5 / Symbian handset would need some type of client). Any thoughts? At $550 per seat looks an expensive way to transfer calls between networks but I've never seen another CPE piece of equipment that can do this. http://www.divitas.com/products Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). [image: Call Button]http://click.mexuar.com/webuser/click/7/userurl/Cognationhttp://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ed Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323friends peer realtime
I have never tried it but I would assume that you can copy the way asterisk works with IAX and SIP. Create your own table in mysql and try editing extconfig.conf and see what happens. - Original Message - From: ~Russell To: asterisk-users@lists.digium.com Sent: Sunday, May 27, 2007 7:50 PM Subject: [asterisk-users] h323friends peer realtime Can anyone help me to make h323friends peer realtime ? -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] execute commands after hangup
It seem's to be exiting after it is set. Try NoOp'ing after you set it.
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 27, 2007 10:36 PM
Subject: [asterisk-users] execute commands after hangup
I have a few commands I wish to run after a hangup.
It looks like only the first 2 commands are run after hangup.
I am using 1.4.3
How can I get the entire loop to run 10 times. ( I know my example just
has noop's but its an example).
exten = h,1,Set(i=1)
exten = h,n,While($[${i} 10])
exten = h,n,Noop(jerry)
exten = h,n,Set(i=$[${i} + 1])
exten = h,n,EndWhile
exten = h,n,Noop(jerry)
The only other item to know is this is a call connected to console/dsp.
Hangup on console
== Spawn extension (default, 1041, 4) exited non-zero on
'SIP/devcentos64_to_bt610tMM-081febf8'
-- Executing [EMAIL PROTECTED]:1]
Set(SIP/devcentos64_to_bt610tMM-081febf8, i=1) in new stack
-- Executing [EMAIL PROTECTED]:2]
While(SIP/devcentos64_to_bt610tMM-081febf8, 1) in new stack
== Spawn extension (default, h, 2) exited non-zero on
'SIP/devcentos64_to_bt610tMM-081febf8'
THanks,
Jerry
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 601 - To not make noise when there is VM
Thanks a lot. Worked like a charm. - Original Message - From: Alvin Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 06, 2007 6:00 PM Subject: Re: [asterisk-users] Polycom 601 - To not make noise when there is VM Google: polycom mwi beep -- http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio The solution given works for me... Alvin Dovid B wrote: Hi Guys, I have some Polycom 601's here. It is super annoying that the phone every so often beeps to let me know that I have a VM. Is there any way to turn that off ? (I just want the red led to blink that there is a VM). Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WiFi SIP phones
I am testing this phone as well. It does have some NAT issues but besides for that it works great. What I like about the phone is that you can use both WIFI and GSM at the same time. This I have not seen on many other phones. Most of them allow you to use the GSM or WIFI at once but not both at the same time. - Original Message - From: Shanon Swafford To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, May 24, 2007 8:10 AM Subject: RE: [asterisk-users] WiFi SIP phones I work for ABP Technology and lurk on this list so I hope I'm not breaking any taboos... ABP is now carrying a dual GSM/Wifi phone. We tested 2 models, 1 had Windows-CE on it. Some reason we only have the Non-CE version public right now. http://www.abptech.com/products/Pirelli/DPL10.html VARs/Resellers/ITSPs/Consultants: http://www.abptech.com/support/qa/index.php?target=become_reseller End Users go here and we'll help you find a place to buy one: http://www.abptech.com/aboutus/find_reseller.php Shanon ABP Technology -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Duncan Turnbull Sent: Wednesday, May 23, 2007 10:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] WiFi SIP phones I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/) Its built to use voip or gsm and is about the same price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's okay and they say WPA is only a firmware load away ;-) , and it has a browser to login if you need to. So far so good and then to some degree I am not sure I would use a wifi only phone again That said wifi voip is still occasionally flaky but I much prefer it to soft clients on the laptop. Cheers Duncan -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, 24 May 2007 2:50 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WiFi SIP phones I travel a lot for work. I frequently find hotels that have wifi, free or otherwise available. But I've yet to find it anywhere near sufficient to support voip applications. At least not good enough to compel me to not use my cell phone. If you have control of the host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise why bother. Has anyone ever seen anyone making a voip call on a wif handset ata public hotspot? While that would score many geek points I doubt it would work in many places. About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously flawed so I resold it after a few months and settled on the Aastra desk phone. I do wish the cordless handsets were a little more like a Panasonic cordless phone...more buttons...easier to program, etc. Michael On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote: On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote: I must say that I've VERY happy with my Aastra 4801 CT phones. I think that they're DECT. Each can have up to six cordless handsets. Technically its a 9 line phone, but if you use G.729 you can only sustain two calls at once. I can have a call on the portable and easily take another on the base. I am also an extremely happy user of an Aastra 480i CT. Awesome phone. However, I was under the impression that the OP was looking for a WiFi phone that could be carried from place to place, but I may be wrong... -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip client registers then unregisters
Have you tried using another phone and compare the results ? - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 16, 2007 2:34 PM Subject: [asterisk-users] Sip client registers then unregisters I have a remote user with Eyebeam on a laptop. Internet connectivity seems good, there is no packet loss to that location from the PBX. Everytime the user starts eyebeam, the application tries to register. Asterisk accepts the registration but the reply never gets to the client application, so it thinks it has not been accepted and times out. Then Asterisk unregisters the extension. -- Registered SIP '881' at 212.248.xxx.xxx port 26605 expires 300 -- Unregistered SIP '881' Anyone got any ideas how to debug and fix this? -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test tools of Asterisk server
I don't know about bandwith consumption but look at sipp (http://sipp.sourceforge.net/) - Original Message - From: khawla khawla To: asterisk-users@lists.digium.com Sent: Saturday, May 26, 2007 10:33 PM Subject: [asterisk-users] test tools of Asterisk server I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. -- Appelez vos amis de PC à PC -- C'EST GRATUIT Essayez-le maintenant ! -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime problem
Hi, I have installed asterisk-1.4.4 and asterisk-addon-1.4.1. I followed every step to configure RealTime but something is not working properly; the warning that I am geting is: WARNING[32709]: config.c:1229 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available WARNING[1359]: config.c:1229 find_engine: Realtime mapping for 'sipusers' found to engine 'mysql', but the engine is not available My cdr_mysql.conf and res_mysql.conf are: /--- cdr_mysql.conf ---/ [global] hostname = localhost dbname = asteriskcdrdb password = passw0rd user = root table = cdr port = 3306 sock = /var/run/mysqld/mysqld.sock userfield = 1 /--- res_mysql.conf ---/ [general] dbhost = localhost dbname = asteriskrealtime dbuser = root dbpass = passw0rd dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock Mysql is working properly, I test it using the parameters included in res_mysql.conf. Is there a way of checking the asterisk conection to the database manually? How can I check that the driver was installed properly from the addonss? Any idea, suggestion? When I had this issue it was becuase I didn't make make install the add on's. Try make clean and then make make install again. Also start asterisk as asterisk -v and see if you see any errors there. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco remote reboot
Is there a way to remote reboot a Cisco 7940 or 7960 phone via some kind of command? The idea is to force a reboot automatically after changing one of the configuration files. Regards, Paul ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zonbu
Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 Nabeel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: May 27, 2007 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zonbu I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco remote reboot
On May 27, 2007, at 5:29 PM, Paul Aviles wrote: Is there a way to remote reboot a Cisco 7940 or 7960 phone via some kind of command? The idea is to force a reboot automatically after changing one of the configuration files. As long as you have telnet access turned on in the config file, you can telnet to them and issue a reboot from there. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zonbu
Looks very close but the next generation on from the Patton (eg cf card, extra usbs and faster cpu) - though with Chinese manufacturing turning out new models on the same form factor could be totally different. Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Sunday, 27 May 2007 5:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Zonbu Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 Nabeel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: May 27, 2007 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zonbu I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco remote reboot
I know that part, I am looking for a way to do it directly from asterisk. Lets say you change a value in the XML configuration of one unit, then from the asterisk box send the signal and reboot only the affected unit. I guess we can always script the telnet process and reboot like that if nothing else. Regards, Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Sunday, May 27, 2007 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco remote reboot On May 27, 2007, at 5:29 PM, Paul Aviles wrote: Is there a way to remote reboot a Cisco 7940 or 7960 phone via some kind of command? The idea is to force a reboot automatically after changing one of the configuration files. As long as you have telnet access turned on in the config file, you can telnet to them and issue a reboot from there. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zonbu
$99,00 for one box, but you need a subscription plan... Zonbu is $99 with a two-year subscription plan. With month to month plan, Zonbu is $249. Sds, Gustavo From: Nabeel Jafferali [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Zonbu Date: Sun, 27 May 2007 17:35:20 -0400 Looks like a rebadged Patton 6075 to me: http://www.patton.com/products/pe_products.asp?category=337 Nabeel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: May 27, 2007 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zonbu I just came across www.Zonbu.com http://www.zonbu.com/ it's a fanless box about the size of a paperback book. It has no hard drive but runs it's Linux OS on a flash card - relying on document storage from an online service (rebadged Amazon S3). http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html I wonder who's going to be the first to hack an asterisk server onto this thing? At $99 it's a hell of an option for a fanless Asterisk server. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Call Button http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Verifique já a segurança do seu PC com o Verificador de Segurança do Windows Live OneCare! http://onecare.live.com/site/pt-br/default.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Please advise: Asterisk on Dell PowerEdge 1750 w/ hyperthreading
I've to setup Asterisk on a Dell PowerEdge 1750 server. Its dual Xeon 3GHz with Hyperthreading. People on this list who have experience with this server please advise me how is the performance of Asterisk on this server, what flavour of linux is good on it etc. Is Hyperthreading going to be a problem or not. I once read somewhere that hyperthreading caused some voice quality problems in Asterisk. Is it fixed in or not yet? Any other suggestions will also be helpful. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Start recording automatically when
1. RE: Start recording automatically when xferring to
anextension? (Don Pobanz)
Message: 1
Date: Fri, 25 May 2007 11:54:33 -0500
From: Don Pobanz [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Start recording automatically when
xferring to anextension?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
J French wrote Friday, May 25, 2007 10:54 AM
I want to start recording the caller automatically when the
receptionist transfers a new sales lead to 567. I don't want
the receptionist to have to press *1 manually for automon.
Can someone recommend how best to accomplish this?
exten = 567,1,Set(CALLERID(name)=SALES CALL)
exten = 567,n,Playback(recorded-for-training)
exten =
Add a couple lines to your 567 extension
exten =
567,n,Set(CALLFILENAME=/var/log/calls/${ARG1}-${CALLERID(num)}-${TIMESTA
MP})
exten = 567,n,MixMonitor(${CALLFILENAME}.wav,b)
567,n,Dial(SIP/phone7SIP/phone8SIP/phone9SIP/phone10SIP/ph
one11SIP/phone12,${SECS_TO_TIMEOUT}) ;Ring Sales Phones
exten = 567,n,Dial(SIP/phone1,${SECS_TO_TIMEOUT}) ;Send back
to Receptionist
Don Pobanz
Hi Don,
What will happen if Receptionist intercom extension 567?? Does the
conversation being records?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reset Polycom phones remotely
Got any rogue DHCP options configured that are set for that MAC address or range? I've noticed when you put in options that the Polycom looks for and have them configured syntactically incorrect, the Polycoms will refuse the entire transaction instead of just the option that was screwed up (timezones with an illegal value in my case). Forum wrote on 5/27/07 12:07 PM: It's definitely ftp. I have given the phone a static ip. When I set it to dhcp it just hangs and cannot get an IP. I can ping the phone and see the web config page so it is on the network. Any more suggestions. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller Sent: Sunday, 27 May 2007 5:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] reset Polycom phones remotely Forum wrote on 5/26/07 5:32 PM: I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I need to reset or reformat the phones - if so how can I do this remotely? Can anyone think of a reason why these 2 phones cannot contact the boot server when the other 2 can? Have you checked their boot server type, and does it match what you have available? If FTP is all you have set up on the boot server and those two phones are set to use TFTP then you would have this issue. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
