[asterisk-users] Users list email totals by year .
2003, 24471 2004, 48608 2005, 59116 2006, 41215 2007, 26414 2008, 20746 2009, 18304 2010, 14948 2011, 11588 2012, 7542 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help determining SpanDSP version
Hello TOm ( all) , ldd -v app_fax.so Should list all items linked against in the module . Hth , JimL On Wed, 26 Jan 2011, Tom Rymes wrote: On 01/25/2011 3:38 PM, Danny Nicholas wrote: [snip] Is there a good way to determine what version of SpanDSP I have installed and whether the app_fax.so module is the same version? [snip] Try these two commands: - whereis spandsp.so - find /|grep spandsp.so Those commands do point towards related pieces, and I think that /usr/include/spandsp/version.h might hold some clues, it doesn't shed any light on the app_fax.so module. Please pardon my ignorance in this area, I'm sure it's straightforward. As for compiling, I have started with a packaged version, and will move to rolling my own as things move along. Many thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing: Anyone have a compiled executable?
Hello Kevin All , On Sat, 16 Jan 2010, Kevin P. Fleming wrote: Doug wrote: app_fax.c from: https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons/trun k/app-spandsp/ Compiled OK: /usr/src/asterisk/app_fax# ls -lta app_fax.* -rwxr-xr-x 1 root root 28869 Jan 13 00:25 app_fax.so -rw-r--r-- 1 root root 25242 Jan 13 00:24 app_fax.c Copied to modules directory: cp -p app_fax.so /usr/lib/asterisk/modules/ There it is: ls -lta /usr/lib/asterisk/modules/app_fax* -rwxr-xr-x 1 root root 28869 Jan 16 02:10 /usr/lib/asterisk/modules/app_fax.so Added a specific line in /etc/asterisk/modules.conf: load = app_fax.so Rebooted. No module loaded: # lsmod | grep fax # app_fax is not a kernel module, it's an Asterisk module. 'lsmod' is never going to show it. Kevin , Sometimes your about as helpful as passing wind . How about telling him howto determine if Asterisk has loaded the module successfully ? Maybe even a grep of /var/log/asterisk/debug or /var/log/asterisk/messages for app_fax . Would have helped him more than that comment . Sorry that reply just really rubbed me wrong . I've found a 'visual only' way of seeing loaded modules under Asterisk 1.4.21.2 ... module reload ?Should show those modules available for reload , So I expect they have been loaded successfully . Here's something that would be good a 'module show loaded' command showing the user the successfully loaded moduels !? Hth , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- 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] Faxing: Anyone have a compiled executable?
Hello Kevin ( All) , On Sat, 16 Jan 2010, Kevin P. Fleming wrote: Mr. James W. Laferriere wrote: Kevin , Sometimes your about as helpful as passing wind . Thanks! Like I said the response just rubbed me wrong , Sorry . How about telling him howto determine if Asterisk has loaded the module successfully ? Users of Asterisk should be able to type 'help' at the Asterisk console prompt, or do Google searches like show asterisk modules. Will show the user a whole bunch of entries or even doing the same search at http://www.voip-info.org/ would probably be better . This would have been a better response . Maybe even a grep of /var/log/asterisk/debug or /var/log/asterisk/messages for app_fax . Would have helped him more than that comment . Sorry that reply just really rubbed me wrong . I've found a 'visual only' way of seeing loaded modules under Asterisk 1.4.21.2 ... module reload ?Should show those modules available for reload , So I expect they have been loaded successfully . Here's something that would be good a 'module show loaded' command showing the user the successfully loaded moduels !? You mean like 'module show'? Or 'module show app_fax.so'? Those commands already exist. No , As far as I can tell . 'modules show' shows you the WHOLE list of available modules NOT just the ones in use . At least that is what appears to be shown when I issue that command line . when I do the 'module reload ?' trick I see those that match my asterisk/*.conf entries . Now as far as the user was concerned the second one mentioned above would have shown him that it was either loaded or not . And either of those lines is what the OP/User was looking for . Twyl , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no mention of an interface associated with this card . Now it is quite possible that Digium in there wisdom has added the necessary components to their drivers that inserts the necessary components into the IF tables thus allowing snmp's IF-MIB to see a known interface . If this is the case where in the driver (or code base) might I find this revelation . I'd sure like to have statistics traps being dumped for this card . (*) 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Tia , JimL 2009/11/27 mickael ropars mrop...@gmail.com: Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael 2009/11/27 mickael ropars mrop...@gmail.com Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- 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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK and SNMP
::zeroDotZero here You have information about interface descryptions, status, speed, type, etc. BR, Micha? 2009/11/27 Mr. James W. Laferriere bab...@baby-dragons.com: Hello Micha ( all) , On Fri, 27 Nov 2009, michal kalinowski wrote: Your Digium card is for linux standard interface like eth0 (ethernet), check IF-MIB.txt and OID from there. BR, Micha? When doing a snmpwalk of the IF-MIB having a (*) installed there is no mention of an interface associated with this card . Now it is quite possible that Digium in there wisdom has added the necessary components to their drivers that inserts the necessary components into the IF tables thus allowing snmp's IF-MIB to see a known interface . If this is the case where in the driver (or code base) might I find this revelation . I'd sure like to have statistics traps being dumped for this card . (*) 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Tia , JimL 2009/11/27 mickael ropars mrop...@gmail.com: Everuthing is working fine, but I have another question to SNMP users: There is no hardware info in the MIB. How can you do to send alarm (when one interface is down for exemple), is there no way to check its status? NB: I am using a Digium card regards Mickael 2009/11/27 mickael ropars mrop...@gmail.com Hi all, I am currently not able to configure SNMP for asterisk, but I am not able to acess to the asterisk MIB (the asterisk MIB is in /usr/share/snmp/mibs/) Does somebody has an example of smnpd.conf file wich is working ? regards Mickael ___ -- 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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ ___ -- 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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 3237 Holden Road | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99709 | only on AXP | +--+ ___ -- 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] OT - DECT SIP Phones
Hello Alan , On Sun, 18 Oct 2009, Alan Lord (News) wrote: On 17/10/09 15:02, --[ UxBoD ]-- wrote: Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) Siemens Gigaset: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Thank you for creating this site keeping up the info available there . But I'd -really- like to see Siemens Data Sheet on the product , Does anyone know where that may (Still|Ever) exist ? One of the most popular posts on my blog over the last 1 1/2 years. It still gets lots of hits from people looking for info on them. FYI We have two sets in our network - they haven't missed a beat since installation. HTH Alan Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Hello John , On Mon, 17 Aug 2009, Lee, John (Sydney) wrote: Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( If you still have the paper work from the box that came to you . The stock agent , if you are lucky , may have written the serial number on the sheet . I have had them do this at various fulfillment centers . Hth , JimL -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, 17 August 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: How to find the serial number ofDigium card? On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote: Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. You cannot. The serial number is not anywhere in the firmware, only on a sticker on the card itself. -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know who's logged in
Hello Mark , On Fri, 27 Mar 2009, Mark Michelson wrote: Mr. James W. Laferriere wrote: On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL Just typing the word agent will result in the message you see. If you press the tab key after typing the word agent you should see that one of your options for completing the command is agent show. This command is definitely in all releases of 1.4. I specifically double-checked and the command works fine for me in 1.4.22. Mark Michelson asterisk-2*CLI help agent No such command 'agent'. asterisk-2*CLI agent No such command 'agent ' (type 'help agent' for other possible commands) Maybe I've mis-configure my compile options or something but ... Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] Know who's logged in
Hello Mark Miquel , On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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-linux 2.1.0.4 released
Hello All , On Tue, 3 Feb 2009, Shaun Ruffell wrote: Thomas Kenyon wrote: On 2/3/2009 17:34, Asterisk Team wrote: The Asterisk development team has released dahdi-linux 2.1.0.4 This release is available for immediate download from http://downloads.digium.com/pub/telephony/dahdi-linux. This release fixes a regression from dahdi-linux 2.1.0 in which it was possible for the kernel to panic when conferencing channels together. Ah, that explains it. :-) Please see http://bugs.digium.com/view.php?id=14183 for more information. The complete change log can be read at: http://downloads.digium.com/pub/telephony/dahdi-linux/releases/ChangeLog-2.1.0.4 Thanks for your continued support of Asterisk! I can't get this to build, the following error is produced: CC [M] /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c: In function 'xproto_get': /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c:96: error: implicit declaration of function 'module_refcount' make[3]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o] Error 1 make[2]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp] Error 2 make[1]: *** [_module_/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.28' make: *** [modules] Error 2 This is on a P4 machine with gcc-4.1.2, (mot sure what else to include really, DAHDI Tools 2.1.0.2, asterisk 1.6.0.3). TIA for any help. is CONFIG_MODULE_UNLOAD defined in your kernel config? And why should this be a build time issue for DAHDI ? Just a ? . Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] snap a number now digium?
Hello John ( All) , On Wed, 21 Jan 2009, John Todd wrote: The SNAP dialer has just been renamed, and it's not only available for AsteriskNow users - it's available for anyone using Asterisk, not just AsteriskNow. The website redirection is not ideal; I agree. We'll try to have it pointed at a specific ADA page shortly, but for the moment the old domain name goes to the digium.com page. Here's a link which contains a location for download of the app, and manuals: http://forums.digium.com/viewtopic.php?t=66048 If it's not working as expected (i.e.: bugs) then you might want to start a discussion of the specifics on the forum board (http://forums.digium.com/viewforum.php?f=26 ) for comments. JT Is there a 'ADA' compatible dialing tool for Linux ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Fri, 19 Dec 2008, Tzafrir Cohen wrote: On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. Are these warning's OK ? Tia , JimL [CC] app_fax.c - app_fax.o app_fax.c:52: warning: no previous prototype for 'ast_tvdiff_sec' app_fax.c:63: warning: no previous prototype for 'ast_tvdiff_us' app_fax.c: In function 'phase_e_handler': app_fax.c:213: warning: implicit declaration of function 't30_get_tx_ident' app_fax.c:213: warning: assignment makes pointer from integer without a cast app_fax.c:214: warning: implicit declaration of function 't30_get_rx_ident' app_fax.c:214: warning: assignment makes pointer from integer without a cast app_fax.c: In function 'set_local_info': app_fax.c:272: warning: implicit declaration of function 't30_set_tx_ident' app_fax.c:276: warning: implicit declaration of function 't30_set_tx_page_header_info' app_fax.c: In function 'transmit_audio': app_fax.c:349: warning: unused variable 'fr' app_fax.c:347: warning: unused variable 'detect_tone' app_fax.c: At top level: app_fax.c:523: warning: 'transmit_t38' defined but not used -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Sun, 21 Dec 2008, Tzafrir Cohen wrote: On Sat, Dec 20, 2008 at 01:50:55PM -0900, Mr. James W. Laferriere wrote: On Fri, 19 Dec 2008, Tzafrir Cohen wrote: On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. Are these warning's OK ? Tia , JimL What version of spandsp do you use? spandsp-0.0.4pre16.tgz Which one is this patch compiling against successfully ? Tho later the make finally blew chuck at LD time ... make[2]: Leaving directory `/home/archive/asterisk/asterisk-1.4.22/main/db1-ast' [LD] abstract_jb.o acl.o aescrypt.o aeskey.o aestab.o alaw.o app.o ast_expr2.o ast_expr2f.o asterisk.o astmm.o astobj2.o audiohook.o autoservice.o callerid.o cdr.o channel.o chanvars.o cli.o config.o cryptostub.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o file.o fixedjitterbuf.o frame.o fskmodem.o global_datastores.o http.o image.o indications.o io.o jitterbuf.o loader.o logger.o manager.o md5.o netsock.o pbx.o plc.o privacy.o rtp.o say.o sched.o sha1.o slinfactory.o srv.o stdtime/localtime.o strcompat.o tdd.o term.o threadstorage.o translate.o udptl.o ulaw.o utils.o editline/libedit.a db1-ast/libdb1.a ../apps/modules.link ../cdr/modules.link ../channels/modules.link ../codecs/modules.link ../formats/modules.link ../funcs/modules.link ../pbx/modules.link ../res/modules.link - asterisk ../apps/app_fax.o: In function `fax_generator_generate': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:319: undefined reference to `fax_tx' ../apps/app_fax.o: In function `load_module': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:793: undefined reference to `span_set_message_handler' ../apps/app_fax.o: In function `phase_e_handler': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:196: undefined reference to `t30_get_transfer_statistics' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:204: undefined reference to `t30_completion_code_to_str' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:206: undefined reference to `t30_completion_code_to_str' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:213: undefined reference to `t30_get_tx_ident' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:214: undefined reference to `t30_get_rx_ident' ../apps/app_fax.o: In function `transmit_audio': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:375: undefined reference to `fax_init' ../apps/app_fax.o: In function `set_logging': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:260: undefined reference to `span_log_set_message_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:261: undefined reference to `span_log_set_level' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:260: undefined reference to `span_log_set_message_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:261: undefined reference to `span_log_set_level' ../apps/app_fax.o: In function `set_file': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:282: undefined reference to `t30_set_tx_file' ../apps/app_fax.o: In function `set_ecm': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:289: undefined reference to `t30_set_ecm_capability' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:290: undefined reference to `t30_set_supported_compressions' ../apps/app_fax.o: In function `transmit_audio': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:386: undefined reference to `fax_set_transmit_on_idle' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:388: undefined reference to `t30_set_phase_e_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:501: undefined reference to `t30_set_phase_e_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:504: undefined reference to `t30_terminate' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:505: undefined reference to `fax_release' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:450: undefined reference to `fax_rx' ../apps/app_fax.o: In function `set_file': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:284: undefined reference to `t30_set_rx_file' ../apps/app_fax.o: In function `set_local_info': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:272: undefined reference to `t30_set_tx_ident' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:276: undefined reference to `t30_set_tx_page_header_info' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make[1]: Leaving
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Sun, 21 Dec 2008, Tzafrir Cohen wrote: On Sat, Dec 20, 2008 at 02:21:28PM -0900, Mr. James W. Laferriere wrote: On Sun, 21 Dec 2008, Tzafrir Cohen wrote: What version of spandsp do you use? spandsp-0.0.4pre16.tgz Which one is this patch compiling against successfully ? 0.0.5pre4 . However with 0.0.4pre16 you should be able to build the agx-addons package mentioned above. For some darned reason , the RxFax would not pickup an inbound fax . So I am trying other options . Your patched app_fax.c . Would 0.0.6pre3 be a problem you think ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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 install dont need download of echo cancel
Hello Tzafir , On Thu, 18 Dec 2008, Tzafrir Cohen wrote: On Thu, Dec 18, 2008 at 12:48:59PM -0500, Matt Watson wrote: after you have configured zaptel manually the first time, copy the menuselect.makeopts file that is generated in the root directory of the zaptel source to a file /etc/zaptel.makeopts. presumably this is available for people that have moved on to DAHDI as well, and I would guess it should be /etc/dahdi.makeopts - but I have not verified that. dahdi-linux does not use menuselect. Then can someone tell me why this file exists ? /home/archive/asterisk/dahdi-linux-complete-2.0.0+2.0.0/tools/menuselect.makeopts # cat !$ MENUSELECT_UTILS=fxstest sethdlc dahdi_diag dahdi_tool MENUSELECT_BUILD_DEPS= MENUSELECT_DEPSFAILED=MENUSELECT_UTILS=sethdlc MENUSELECT_DEPSFAILED=MENUSELECT_UTILS=dahdi_tool Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- 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] Upgrade to Asterisk 1.4 - it's one year's old!
Hello All , On Sat, 15 Dec 2007, Johansson Olle E wrote: Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ The one item mentioned in some of the responses to the thread that this message started is the modification of commands (dialplan others) , variables and such . Tilghman mentioned these changes are collected in UPGRADE.txt . But (I have to admit IMO) , The procedure necessary to follow to get a system running 1.4 is not a upgrade path . It is a migration . ie: duplicate the system(s) running 1.2 successfully today onto seperate hardware make the changes necessary to create a (near as possible) functioning system as the present systems then swap them . If this path was more of a true upgrade path then 1.4 would probably be used far more than 1.2 . Hth , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | [EMAIL PROTECTED] | Fairbanks, AK. 99701 | only on AXP | +--+ ___ --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] DS3 Interface
Hello Gentleman Ladies , On Tue, 9 Oct 2007, Tilghman Lesher wrote: On Tuesday 09 October 2007 14:20:33 Brian West wrote: I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. I'm not exactly sure that you're the right person to be taking offense at someone dragging a project's name through the mud. Please , step back form the keyboard , take a deep breath . then maybe we can get on with the discussion of creating a driver under aterisk for a ds3 card . Tia , JimL -- +-+ | James W. Laferriere | System Techniques | Give me VMS | | NetworkEngineer | 663 Beaumont Blvd | Give me Linux | | [EMAIL PROTECTED] | Pacifica, CA. 94044 | only on AXP | +-+ ___ --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] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???
Hello All , I'd usually just take the latest timestamped tarballs use them , But this has gotten me a tad setback . I want to build astersik-1.4.1 I am not sure which of these is going to work correctly . Anyone else have a better idea than me ? Rsvp , Tia , JimL -rw-r--r--1 00 9397296 Feb 21 01:07 asterisk-core-sounds-es-g722-1.4.6.tar.gz -rw-r--r--1 00 2129399 Feb 21 01:07 asterisk-core-sounds-es-gsm-1.4.6.tar.gz -rw-r--r--1 0012219353 Feb 21 01:07 asterisk-core-sounds-en-alaw-1.4.6.tar.gz -rw-r--r--1 00 7167005 Feb 21 01:07 asterisk-core-sounds-fr-g722-1.4.6.tar.gz -rw-r--r--1 00 6874032 Feb 21 01:07 asterisk-core-sounds-en-g729-1.4.6.tar.gz -rw-r--r--1 00 1623436 Feb 21 01:07 asterisk-core-sounds-fr-gsm-1.4.6.tar.gz -rw-r--r--1 0018603291 Feb 21 01:07 asterisk-core-sounds-es-wav-1.4.6.tar.gz -rw-r--r--1 0012278506 Feb 21 01:07 asterisk-core-sounds-en-ulaw-1.4.6.tar.gz drwxr-xr-x3 008192 Feb 22 00:32 . -rw-r--r--1 0027839721 Feb 22 00:32 asterisk-extra-sounds-en-wav-1.4.5.tar.gz -rw-r--r--1 001375 Feb 22 00:32 asterisk-extra-sounds-en-ulaw-1.4.5.tar.gz -rw-r--r--1 0013675929 Feb 22 00:32 asterisk-extra-sounds-en-g722-1.4.5.tar.gz -rw-r--r--1 00 3235653 Feb 22 00:32 asterisk-extra-sounds-en-gsm-1.4.5.tar.gz -rw-r--r--1 0013473844 Feb 22 00:32 asterisk-extra-sounds-en-alaw-1.4.5.tar.gz -rw-r--r--1 00 2017747 Feb 22 00:32 asterisk-extra-sounds-en-g729-1.4.5.tar.gz drwxr-xr-x4 004096 Feb 22 00:40 .. drwxr-xr-x6 004096 Mar 06 00:50 .svn ncftp ...ephony/sounds/releases dir -alrt -- +-+ | James W. Laferriere | System Techniques | Give me VMS | | NetworkEngineer | 663 Beaumont Blvd | Give me Linux | | [EMAIL PROTECTED] | Pacifica, CA. 94044 | only on AXP | +-+ ___ --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] netstats like command for sip , Is there one ?
Hello All , Is there a command or set of commands that will give the same data resources as 'iax2 show netstats' for sip ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3600 14th Ave SE #20-103 | Give me Linux | | [EMAIL PROTECTED] | Olympia , WA. 98501 | only on AXP | +--+ ___ --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] AEL2
Hello All , On Fri, 9 Jun 2006, Gonzalo Servat wrote: On 6/9/06, Joshua Colp [EMAIL PROTECTED] wrote: [..snip..] I'd just like to note that AEL2 was brought over into Asterisk trunk (what will become 1.4) and the old AEL removed. That's where most development is taking place on AEL2, and why you don't see patches on the bug tracker. Hi Joshua, I was just reading the bug report and noticed it has been merged. Awesome news! I'm still using 1.2.x so sticking to AEL for now, but I'm going to quickly move to AEL2 as soon as I upgrade to 1.4! (whenever it comes out) Regards, Gonzalo. Something along the same lines . Does anyone have link to complete documentation of AEL2 ? Did a small bit of checking nothing that seemed complete . Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3600 14th Ave SE #20-103 | Give me Linux | | [EMAIL PROTECTED] | Olympia , WA. 98501 | only on AXP | +--+ ___ --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 on laptop connected to POTS line
Hello All , On Thu, 2 Feb 2006, [EMAIL PROTECTED] wrote: Original Message Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS line From: Tzafrir Cohen [EMAIL PROTECTED] Date: Thu, February 02, 2006 9:15 am To: asterisk-users@lists.digium.com On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote: Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Look at Xorcom's USB channel Bank. Which is a great product and you should all get one (and the fact that I'm a Xorcom employee has nothing to do with this recommendation), but sadly, still lacks FXO ports. If Xorcom could offer something similar with 2-4 FXOs I'd just have to buy at least one. Heck of an idea for a product, a quad FXO adapter interfaced to Asterisk via local USB port. Wow! If one could get this in 1-3 FXO 1-3FXS ports(*) in an apropriate combination ... Where the USER can select which combo s/he wants at home , Not by buying a hardwired device . Then that would be something to buy . (*) 1FXO 3FXS , 2FXO 2FXS , 3FXO 1FXS . My $.02 , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | | http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr | +--+ ___ --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 always uses 127.0.0.1 address
Hello All , On Sat, 21 Jan 2006, Alberto Sagredo wrote: Maybe you have not configured correcly your sip.conf externip=your_external_ip try this RumaTech escribió: Something right down this alley . What happens if I have more than one interface I want asterisk to listen on ? 1 ) What is the syntax of externip= for two (or more) ? 2 ) What is the syntax of bindaddress= for two (or more) ? 3 ) What is the proper method to define the proper IP's or interfaces to use ? Tia , JimL Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as [EMAIL PROTECTED] (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID(SIP/phone2-22c3, CID Name CIDNUMBER) in new stack -- Executing Dial(SIP/phone2-22c3, SIP/sipnet/84959741926) in new stack We're at 127.0.0.1 port 18900 ANy help is appreciated, Thanks, Rudolf -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | | http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr | +--+___ --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 1.2.2 Released!
Hello Announce All , On Wed, 18 Jan 2006, Asterisk Development Team wrote: Greetings everyone! The 1.2.2 versions of Asterisk, Zaptel, and Libpri have now been released. The source tarballs are available for download on ftp.digium.com. For details about what has changed, see the ChangeLog for Asterisk, Zaptel, or Libpri. We are also excited to announce the release of a special version of Asterisk 1.2.2, called Asterisk-NetSec. It includes some very exciting features not available in any other version of Asterisk, or even any other related product! Please view the appropriate README and ChangeLog for more details. Asterisk-addons and Asterisk-sounds will remain at version 1.2.1. Previously, all packages were updated to reflect a matching version number, even if no changes have been made. From now on, releases will only be made when changes have actually been made. Even if version numbers do not match, it is safe to use all of these releases together, as long as all of them are the latest version available. Thank you! The one thing that annoys me most is a announcment with out a url: to what it is announcing . Can we please correct this ? Tia , JimL ps: Not that I can't find it , but ... is just courtisy to others . -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | | http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr | +--+ ___ --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] Bind asterisk to multiple IPs (reply problem)
Hello Kevin , On Thu, 5 Jan 2006, Kevin P. Fleming wrote: Ales Vizdal, AVONET, s.r.o. wrote: I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0 (ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA registers to a.b.c.e, asterisk sends register response from a.b.c.d and client ignores reply, because a.b.c.e != a.b.c.d. Is it bug, feature or some kind of misconfiguration? It's a known bug. It is being worked on, but the results won't be in an Asterisk release until 1.4. Is the in developement functionality in the svn ? Ie: can I do .. svn update http://svn.digium.com/svn/asterisk/trunk asterisk to acquire it ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --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] zapata directory not found in svn .
Hello Kevin , On Tue, 29 Nov 2005, Kevin P. Fleming wrote: Mr. James W. Laferriere wrote: Hello All , no zapata diredtory , tho zaptel README says many of the testing programs require its libraries . Please enlighten me . Tia , JimL The zapata directory was not imported into SVN. If anything actually does need it, you can get it from CVS. Any reason why ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --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] zapata directory not found in svn .
Hello All , no zapata diredtory , tho zaptel README says many of the testing programs require its libraries . Please enlighten me . Tia , JimL $ svn checkout http://svn.digium.com/svn/zapata/trunk zapata svn: PROPFIND request failed on '/svn/zapata/trunk' svn: Could not open the requested SVN filesystem From zaptel/README ... ..snip.. Requirements: Some of the testing programs still require the zapata library The zttool program requires libnew ..snip.. -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --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-rc1 and sip show inuse
Hello Kevin , On Wed, 16 Nov 2005, Kevin Hanson wrote: Steven Ringwald wrote: I apologize if this question has been asked before. Did something change with the behaviour of the 'sip show inuse' command between 1.0.9 and 1.2-rc1? I used to be able to see a list of extensions and the number of in/out calls. Now it just reports: asterisk*CLI sip show inuse * User name In use Limit * Peer name In use Limit no matter how many calls are being used. asterisk*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.70.128 1234339ad96826e 00102/0 ulaw No Tx: ACK 192.168.70.116 1235723e1612-52 00101/2 ulaw No Rx: ACK 2 active SIP channels Any info about getting the previous functionality back would be greatly appreciated. Steve I think you have to have call-limit set in sip.conf. I had the same problem, then set call-limit=10 and 'sip show inuse' worked. Did that on a cvs pull ... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-10-31 23:09:39 UTC I get the same response from 'inuse' as Steven does even with the addition of the 'call-limit=' in sip.conf . Any other suggestions ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clarification on chan_modem.so module
Hello BJ all , On Thu, 10 Nov 2005, BJ Weschke wrote: On 11/10/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Just so I am clear for version 1.2 has chan_modem.so been depreciated? That means I should also remove this module from loading in the modules.conf if I am using Asterisk 1.2 rc1. Do I have to do anything to replace this functionality (I do not really understand what chan_modem.so was used for other than it seemed to be linked to musiconhold...) Yes. It has been deprecated. I believe it's original purpose was to be able to use the voice modems out there as FXO ports in Asterisk. You musiconhold will function without it. Can you speak to , what other functionality has taken it place ? Some people out here use the chan_modem.so functionality . Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Include statement options docs .
Hello All , Can someone point me to a full description of all options allowed with the include statement ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] timed allow functionality of 'include ='s
Hello All , Been looking at the timed allow functionality of the 'include =' statements . Without docs on the functionality I am plain guessing about the syntax format . I am trying to Allow a context the ability to dial out of my system at a time after local business hours . My best guess at a proper allow usage is below . But I am unsure if the two seperate instances will effect calls already in progress ? And if there might be a single line version of this ? Tia , JimL include = dial-out|18:00-23:59|sun-sat|* include = dial-out|00:00-07:00|sun-sat|* -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time based call direction
Hello Kyle All , On Wed, 2 Nov 2005, Kyle Hagan wrote: Adam Moffett wrote: include = atlunchcontext|11:00-11:59|mon-fri|* include = notatlunchcontext|09:00-10:59|mon-fri|* include = notatlunchcontext|12:00-18:00|mon-fri|* include = afterhourscontext|18:01--8:59|mon-fri|* I wasn't aware that include allowed a time qualifier. Does that mean that the specified context will only be included at the specified time? Correct. We have been useing that here for some time now. Can someone point me to a full description off any all options allowed with the include statement ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timestamps in Console?
Hello Jorge All , On Mon, 31 Oct 2005, Jorge Merlino wrote: There is the -T option when running the CLI but I think it only works in 1.2 Regards Jorge El Lun 31 Oct 2005 12:30, [EMAIL PROTECTED] escribió: Hello! Lately, I've been keeping a close eye on an Asterisk box by staying logged into the console for long periods of time. However, it can be very difficult to know how long a telephone call lasts when this is all you see: -- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in new stack -- Called g2/Number -- Zap/5-1 answered SIP/SIP105-8e34 -- Hungup 'Zap/5-1' Did that telephone call last only a few seconds because there was a problem, or a few minutes because there wasn't? It's impossible to tell. Is there a way to add timestamps to each line in the console so you know exactly how long a call took? Or is there another way of telling directly within the console? Thank you very much! Tim Massey As you can see below this option does not put time stamps on the reports from actions in the dialplan . Can this option be extended to the operations within the dialplan ? Tia , JimL [EMAIL PROTECTED]:~# asterisk -Trn [Oct 31 14:25:35] Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. [Oct 31 14:25:35] Written by Mark Spencer [EMAIL PROTECTED] [Oct 31 14:25:35] = [Oct 31 14:25:35] Connected to Asterisk CVS-HEAD currently running on asterisk-1 (pid = 240) Verbosity is at least 3 -- Remote UNIX connection -- Executing Macro(SIP/2701-51eb, calluser|2702|30) in new stack -- Executing Dial(SIP/2701-51eb, SIP/2702|30|Tt) in new stack -- Called 2702 -- SIP/2702-21d9 is ringing asterisk-1*CLI -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Zaptel Versions Command?
Hello Bart ,
On Mon, 31 Oct 2005, Bart Fisher wrote:
Thanks, but what I was really hoping for was something that could be used in a
script to report current revisions... me sad
Bart
- Original Message - From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 31, 2005 12:33 PM
Subject: Re: [Asterisk-Users] Asterisk and Zaptel Versions Command?
Yeah, show versions in the CLI will give you the version of your asterisk
build
Also you can do the following in the CLI:
show version files filename
where filename is a valid file name.
As always in Linux you can press TAB to get a list of available
commands in the CLI, for example you can type:
show version files {TAB}
that will give you a list of all the files you can then type the file
you want. Or you could narrow it down like this:
show version files chan{TAB}
that will give you a list of all the avaiable files that start with
chan, you could also do just {TAB} to get a list of all the commands.
To get help you could type help command.
Hope this helps.
On 10/31/05, Bart Fisher [EMAIL PROTECTED] wrote:
Is there a command line for discovery of Asterisk and Zaptel Versions?
Bart
Give this a try . Hth , JimL
[EMAIL PROTECTED]:~# asterisk -rx show version
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-06 16:59:09 UTC
-- Remote UNIX connection
--
+--+
| James W. Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux |
| [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP |
+--+
___
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Re: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello All , On Fri, 28 Oct 2005, Mr. James W. Laferriere wrote: On Thu, 27 Oct 2005, Mr. James W. Laferriere wrote: On Thu, 27 Oct 2005, Phil Pritchard wrote: only new to asterisk, but have had some hardware exp. stay away from irq9 its tied to irq2 and will always be shared, Paul has the go.. in bios disable serial and or usb (if not using) and make sure irda is not enabled. another one is the lpt port if your not using that, there is another irq you can steel.. ALL I mean all serial/parrallel/...'everything I can find'... has been turned off in the bios . And I have recompiled a kernel with those same items turned off in it . That d??ned module wants to load at irq 9 no matter what I do . Of course there is no way to set irq's to a particular pci slot in the bios . Does anyone now howto set irq say at the boot: or in modprobe.conf ? dont share interrupts, as a rule(if you can help it)... it usually leads to system instability and usually under load. Quite well understand this point . Have heard it on this list many times . And am doing my best NOT too . UBCD ...(www.ultimatebootcd.com). has some nice tools that can probe a system to give a second appinion on interrupt conflicts, ram and hard drive errors. its my best tool for hardware problems.. IMO , The mirrors have the su??iest download schemes I have seen in some time .\IMO I have yet to burn that image but as soon as I do I'll boot it on that piece of junk I bought for near next to nothing . Which is almost what it is worth , Nothing . Thank you for your input , Every bit helps . JimL Finally got that da??ed wcfxo to load on a irq by itself (*). Had to turn off the last item of the onbord devices the ether buy an ether card to get connectivity . But even with the suggestion by 'Paul' to use a two line cord finally using a singular irq , The config's I sent last time have not changed . The x100p/wcfxo combination see the line ringing (**) . But asterisk does NOT see it on the console nor does it pick up the line . Quite frustrating when everything should be ok per every example I've seen still nothing positive to show for it . ANY suggestions/questions/... Please pipe up . Tia , JimL For everybodies info , Make sure that there isn't an entry like ... noload = chan_zap.so in /etc/asterisk/modules.conf . That was what the problem was all along . Tnx to all who helped . JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello All ,
On Thu, 27 Oct 2005, Mr. James W. Laferriere wrote:
On Thu, 27 Oct 2005, Phil Pritchard wrote:
only new to asterisk, but have had some hardware exp.
stay away from irq9 its tied to irq2 and will always be shared, Paul has
the go.. in bios disable serial and or usb (if not using) and make sure irda
is not enabled. another one is the lpt port if your not using that, there is
another irq you can steel..
ALL I mean all serial/parrallel/...'everything I can find'... has
been
turned off in the bios . And I have recompiled a kernel with those
same
items turned off in it . That d??ned module wants to load at irq 9 no
matter what I do . Of course there is no way to set irq's to a
particular pci slot in the bios .
Does anyone now howto set irq say at the boot: or in modprobe.conf ?
dont share interrupts, as a rule(if you can help it)... it usually leads to
system instability and usually under load.
Quite well understand this point . Have heard it on this list many
times . And am doing my best NOT too .
UBCD ...(www.ultimatebootcd.com). has some nice tools that can probe a
system
to give a second appinion on interrupt conflicts, ram and hard drive
errors.
its my best tool for hardware problems..
IMO , The mirrors have the su??iest download schemes I have seen in
some time .\IMO
I have yet to burn that image but as soon as I do I'll boot it on that
piece of junk I bought for near next to nothing . Which is almost what
it is worth , Nothing .
Thank you for your input , Every bit helps . JimL
Finally got that da??ed wcfxo to load on a irq by itself (*). Had to
turn off the last item of the onbord devices the ether buy an ether
card to get connectivity . But even with the suggestion by 'Paul' to
use a two line cord finally using a singular irq , The config's I
sent last time have not changed . The x100p/wcfxo combination see the
line ringing (**) . But asterisk does NOT see it on the console nor
does it pick up the line . Quite frustrating when everything should be
ok per every example I've seen still nothing positive to show for it .
ANY suggestions/questions/... Please pipe up . Tia , JimL
(*)
Oct 28 09:43:47 asterisk-test kernel: Zapata Telephony Interface Registered on
major 196
Oct 28 09:43:47 asterisk-test kernel: PCI: Found IRQ 5 for device :01:01.0
Oct 28 09:43:47 asterisk-test kernel: Registered Span 1 ('WCFXO/0') with 1
channels
Oct 28 09:43:47 asterisk-test kernel: Span ('WCFXO/0') is new master
Oct 28 09:43:47 asterisk-test kernel: New regoffset: 7
Oct 28 09:43:47 asterisk-test kernel: wcfxo: DAA mode is 'FCC'
Oct 28 09:43:47 asterisk-test kernel: Found a Wildcard FXO: Wildcard X101P
Oct 28 09:43:47 asterisk-test kernel: BATTERY!
Oct 28 09:43:47 asterisk-test kernel: Registered tone zone 0 (United States /
North America)
(**)
Oct 28 09:46:35 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:37 asterisk-test kernel: wcfxo: NO RING!
Oct 28 09:46:41 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:43 asterisk-test kernel: wcfxo: NO RING!
Oct 28 09:46:47 asterisk-test kernel: wcfxo: RING!
Oct 28 09:46:49 asterisk-test kernel: wcfxo: NO RING!
--
+--+
| James W. Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux |
| [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP |
+--+
___
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Re: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello Phil , On Thu, 27 Oct 2005, Phil Pritchard wrote: only new to asterisk, but have had some hardware exp. stay away from irq9 its tied to irq2 and will always be shared, Paul has the go.. in bios disable serial and or usb (if not using) and make sure irda is not enabled. another one is the lpt port if your not using that, there is another irq you can steel.. ALL I mean all serial/parrallel/...'everything I can find'... has been turned off in the bios . And I have recompiled a kernel with those same items turned off in it . That d??ned module wants to load at irq 9 no matter what I do . Of course there is no way to set irq's to a particular pci slot in the bios . Does anyone now howto set irq say at the boot: or in modprobe.conf ? dont share interrupts, as a rule(if you can help it)... it usually leads to system instability and usually under load. Quite well understand this point . Have heard it on this list many times . And am doing my best NOT too . UBCD ...(www.ultimatebootcd.com). has some nice tools that can probe a system to give a second appinion on interrupt conflicts, ram and hard drive errors. its my best tool for hardware problems.. IMO , The mirrors have the su??iest download schemes I have seen in some time .\IMO I have yet to burn that image but as soon as I do I'll boot it on that piece of junk I bought for near next to nothing . Which is almost what it is worth , Nothing . Thank you for your input , Every bit helps . JimL Mr. James W. Laferriere wrote: Hello Paul all , On Wed, 26 Oct 2005, Mr. James W. Laferriere wrote: Hello Paul all , I've tried everything I know to attempt to get the wcfxo.ko not to use irq 9 . THe 6 line cord does not appear to effect the signaling to the x100p card , I have turned up the debugging have that being syslog'd . Have debugging on zaptel as well . Nothing seems out of the ordinary . But monitoring from 'asterisk -d -v -nr'console does not show anything '.' . Have I forgotten some configurations or magical incantation ? Tia , JimL On Wed, 26 Oct 2005, Paul wrote: First I don't like the 6 line cord. Use an rj11 2 wire cord, but watch the crossover vrs straight on the old red and green. Next the interrupt must be fixed. Do this in the CMOS before you boot. Go to the PCI bus assignments and set the IRQ or go and disable the serial ports thereby allowing irq 3 and 4 to be assigned. :) Paul Sorry about the top posting ... Also forgot the syslog output . Tia , JimL Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: subsystem zaptel: registering Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: class Oct 26 20:11:19 asterisk-test kernel: kobject zaptimer: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapchannel: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zappseudo: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapctl: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: Zapata Telephony Interface Registered on major 196 Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: drivers Oct 26 20:11:19 asterisk-test kernel: PCI: Found IRQ 9 for device :01:02.0 Oct 26 20:11:19 asterisk-test kernel: PCI: Sharing IRQ 9 with :00:1f.3 Oct 26 20:11:19 asterisk-test kernel: kobject zap1: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: New regoffset: 7 Oct 26 20:11:20 asterisk-test kernel: wcfxo: DAA mode is 'FCC' Oct 26 20:11:20 asterisk-test kernel: Found a Wildcard FXO: Wildcard X101P Oct 26 20:11:20 asterisk-test kernel: Recalculating slaves on WCFXO/0/0 Oct 26 20:11:20 asterisk-test kernel: Done Recalculating slaves on WCFXO/0/0 (last is WCFXO/0/0) Oct 26 20:11:20 asterisk-test kernel: Configured channel WCFXO/0/0, flags 0201, sig 2004 Oct 26 20:11:20 asterisk-test kernel: Registered tone zone 0 (United States / North America) Oct 26 20:11:20 asterisk-test kernel: BATTERY! Oct 26 20:11:39 asterisk-test kernel: Out of storage space Oct 26 20:11:48 asterisk-test kernel: RING! Oct 26 20:11:50 asterisk-test kernel: NO RING! Oct 26 20:11:54 asterisk-test kernel: RING! Oct 26 20:11:56 asterisk-test kernel: NO RING! Oct 26 20:13:50 asterisk-test kernel: RING! Oct 26 20:13:52 asterisk-test kernel: NO RING! Oct 26 20:13:56
[Asterisk-Users] x100p (FXO) not being seen by asterisk (is my best guess) .
Hello All , I installed a x100p clone device in a system . I pulled (Wed, 26 Oct 2005 early am) cvs compiled installed for a 2.6 kernel . We call this system 'test' . This system is a HP Vectra with a 933MHZ processor . Test has all the z* modules installed (*1). and can see the x100p (ie *2) . The other system has been operational for some time . I have a ata-186 responding to two numbers 21 22 . I attached a 6 line cord between the ata-186 port 22 the 'line' side of the x100p . I have configured the 3 necessary files (*3) . Now when I call 22 from an ipphone the x100p does not pickup the line . All it does is ring . I have to be missing something here . Tia , JimL ps: Here is one problem I have yet been unable to resolv . I am unable to move the x100p to some other IRQ at present it is sharing irq 9 with the SMBus . Anyone know how to work around this ? 00:1f.3 SMBus: Intel Corporation 82801AA SMBus (rev 02) Subsystem: Intel Corporation 82801AA SMBus Flags: medium devsel, IRQ 9 I/O ports at 1810 [size=16] 01:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 64, IRQ 9 I/O ports at 2000 [size=256] Memory at ec10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 (*1) # modprobe -vl /lib/modules/2.6.13.2/misc/ztdynamic.ko /lib/modules/2.6.13.2/misc/ztdummy.ko /lib/modules/2.6.13.2/misc/ztd-loc.ko /lib/modules/2.6.13.2/misc/ztd-eth.ko /lib/modules/2.6.13.2/misc/zaptel.ko /lib/modules/2.6.13.2/misc/wcusb.ko /lib/modules/2.6.13.2/misc/wcte11xp.ko /lib/modules/2.6.13.2/misc/wctdm.ko /lib/modules/2.6.13.2/misc/wct4xxp.ko /lib/modules/2.6.13.2/misc/wct1xxp.ko /lib/modules/2.6.13.2/misc/wcfxo.ko /lib/modules/2.6.13.2/misc/torisa.ko /lib/modules/2.6.13.2/misc/tor2.ko /lib/modules/2.6.13.2/misc/pciradio.ko (*2) # cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 FXSKS (*3) # cat /etc/zaptel.conf fxsks=1 loadzone=us defaultzone=us # cat /etc/asterisk/zapata.conf [trunkgroups] [channels] ;language=en signalling=fxs_ks context=x100p-incoming ;usecallerid=yes ;hidecallerid=no ;callwaiting=no ;threewaycalling=yes ;transfer=yes ;echocancel=yes ;echotraining=yes channel = 1 # cat /etc/asterisk/zapata.conf ...snip... [x100p-incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,2,Echo -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello Paul all , I've tried everything I know to attempt to get the wcfxo.ko not to use irq 9 . THe 6 line cord does not appear to effect the signaling to the x100p card , I have turned up the debugging have that being syslog'd . Have debugging on zaptel as well . Nothing seems out of the ordinary . But monitoring from 'asterisk -d -v -nr' console does not show anything '.' . Have I forgotten some configurations or magical incantation ? Tia , JimL On Wed, 26 Oct 2005, Paul wrote: First I don't like the 6 line cord. Use an rj11 2 wire cord, but watch the crossover vrs straight on the old red and green. Next the interrupt must be fixed. Do this in the CMOS before you boot. Go to the PCI bus assignments and set the IRQ or go and disable the serial ports thereby allowing irq 3 and 4 to be assigned. :) Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mr. James W. Laferriere Sent: Wednesday, October 26, 2005 7:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) . Hello All , I installed a x100p clone device in a system . I pulled (Wed, 26 Oct 2005 early am) cvs compiled installed for a 2.6 kernel . We call this system 'test' . This system is a HP Vectra with a 933MHZ processor . Test has all the z* modules installed (*1). and can see the x100p (ie *2) . The other system has been operational for some time . I have a ata-186 responding to two numbers 21 22 . I attached a 6 line cord between the ata-186 port 22 the 'line' side of the x100p . I have configured the 3 necessary files (*3) . Now when I call 22 from an ipphone the x100p does not pickup the line . All it does is ring . I have to be missing something here . Tia , JimL ps: Here is one problem I have yet been unable to resolv . I am unable to move the x100p to some other IRQ at present it is sharing irq 9 with the SMBus . Anyone know how to work around this ? 00:1f.3 SMBus: Intel Corporation 82801AA SMBus (rev 02) Subsystem: Intel Corporation 82801AA SMBus Flags: medium devsel, IRQ 9 I/O ports at 1810 [size=16] 01:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 64, IRQ 9 I/O ports at 2000 [size=256] Memory at ec10 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 (*1) # modprobe -vl /lib/modules/2.6.13.2/misc/ztdynamic.ko /lib/modules/2.6.13.2/misc/ztdummy.ko /lib/modules/2.6.13.2/misc/ztd-loc.ko /lib/modules/2.6.13.2/misc/ztd-eth.ko /lib/modules/2.6.13.2/misc/zaptel.ko /lib/modules/2.6.13.2/misc/wcusb.ko /lib/modules/2.6.13.2/misc/wcte11xp.ko /lib/modules/2.6.13.2/misc/wctdm.ko /lib/modules/2.6.13.2/misc/wct4xxp.ko /lib/modules/2.6.13.2/misc/wct1xxp.ko /lib/modules/2.6.13.2/misc/wcfxo.ko /lib/modules/2.6.13.2/misc/torisa.ko /lib/modules/2.6.13.2/misc/tor2.ko /lib/modules/2.6.13.2/misc/pciradio.ko (*2) # cat /proc/zaptel/1 Span 1: WCFXO/0 Wildcard X101P Board 1 1 WCFXO/0/0 FXSKS (*3) # cat /etc/zaptel.conf fxsks=1 loadzone=us defaultzone=us # cat /etc/asterisk/zapata.conf [trunkgroups] [channels] ;language=en signalling=fxs_ks context=x100p-incoming ;usecallerid=yes ;hidecallerid=no ;callwaiting=no ;threewaycalling=yes ;transfer=yes ;echocancel=yes ;echotraining=yes channel = 1 # cat /etc/asterisk/zapata.conf ...snip... [x100p-incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,2,Echo -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello Paul all , On Wed, 26 Oct 2005, Mr. James W. Laferriere wrote: Hello Paul all , I've tried everything I know to attempt to get the wcfxo.ko not to use irq 9 . THe 6 line cord does not appear to effect the signaling to the x100p card , I have turned up the debugging have that being syslog'd . Have debugging on zaptel as well . Nothing seems out of the ordinary . But monitoring from 'asterisk -d -v -nr' console does not show anything '.' . Have I forgotten some configurations or magical incantation ? Tia , JimL On Wed, 26 Oct 2005, Paul wrote: First I don't like the 6 line cord. Use an rj11 2 wire cord, but watch the crossover vrs straight on the old red and green. Next the interrupt must be fixed. Do this in the CMOS before you boot. Go to the PCI bus assignments and set the IRQ or go and disable the serial ports thereby allowing irq 3 and 4 to be assigned. :) Paul Sorry about the top posting ... Also forgot the syslog output . Tia , JimL Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: subsystem zaptel: registering Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: class Oct 26 20:11:19 asterisk-test kernel: kobject zaptimer: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapchannel: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zappseudo: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapctl: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: Zapata Telephony Interface Registered on major 196 Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: drivers Oct 26 20:11:19 asterisk-test kernel: PCI: Found IRQ 9 for device :01:02.0 Oct 26 20:11:19 asterisk-test kernel: PCI: Sharing IRQ 9 with :00:1f.3 Oct 26 20:11:19 asterisk-test kernel: kobject zap1: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: New regoffset: 7 Oct 26 20:11:20 asterisk-test kernel: wcfxo: DAA mode is 'FCC' Oct 26 20:11:20 asterisk-test kernel: Found a Wildcard FXO: Wildcard X101P Oct 26 20:11:20 asterisk-test kernel: Recalculating slaves on WCFXO/0/0 Oct 26 20:11:20 asterisk-test kernel: Done Recalculating slaves on WCFXO/0/0 (last is WCFXO/0/0) Oct 26 20:11:20 asterisk-test kernel: Configured channel WCFXO/0/0, flags 0201, sig 2004 Oct 26 20:11:20 asterisk-test kernel: Registered tone zone 0 (United States / North America) Oct 26 20:11:20 asterisk-test kernel: BATTERY! Oct 26 20:11:39 asterisk-test kernel: Out of storage space Oct 26 20:11:48 asterisk-test kernel: RING! Oct 26 20:11:50 asterisk-test kernel: NO RING! Oct 26 20:11:54 asterisk-test kernel: RING! Oct 26 20:11:56 asterisk-test kernel: NO RING! Oct 26 20:13:50 asterisk-test kernel: RING! Oct 26 20:13:52 asterisk-test kernel: NO RING! Oct 26 20:13:56 asterisk-test kernel: RING! Oct 26 20:13:57 asterisk-test kernel: NO RING! -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fyi: I-D ACTION:draft-guy-enumiax-00.txt (fwd)
Hello ALl , Tis about time !-) . JimL -- Forwarded message -- Date: Thu, 20 Oct 2005 15:50:02 -0400 From: [EMAIL PROTECTED] To: i-d-announce@ietf.org Subject: I-D ACTION:draft-guy-enumiax-00.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. Title : IANA Registration for IAX Enumservice Author(s) : E. Guy Filename: draft-guy-enumiax-00.txt Pages : 11 Date: 2005-10-20 This document registers the IAX2 Enumservice using the URI scheme 'iax2:' as per the IANA registration process defined in the ENUM specification RFC3761. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-guy-enumiax-00.txt To remove yourself from the I-D Announcement list, send a message to [EMAIL PROTECTED] with the word unsubscribe in the body of the message. You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the username anonymous and a password of your e-mail address. After logging in, type cd internet-drafts and then get draft-guy-enumiax-00.txt. A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: [EMAIL PROTECTED] In the body type: FILE /internet-drafts/draft-guy-enumiax-00.txt. NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the mpack utility. To use this feature, insert the command ENCODING mime before the FILE command. To decode the response(s), you will need munpack or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with multipart MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ftp://ftp/internet-drafts/draft-guy-enumiax-00 ___ I-D-Announce mailing list I-D-Announce@ietf.org https://www1.ietf.org/mailman/listinfo/i-d-announce ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
Hello Leif , The appendices A B are missing from the zip file available at the location mentioned below . Is there some reason of copyright that is not mentioned here ? Tia , JimL On Sat, 15 Oct 2005, Leif Madsen wrote: Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on Read the book online! for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for those new to Asterisk, and to give back to a project that has given us all so much. I would personally like to thank Jared Smith, Jim van Meggelen, Michael Loukides (our editor) and the entire O'Reilly Media staff. The book is currently shipping, and should be available at all major book stores in paperback, and also online from http://www.oreilly.com/catalog/asterisk/ and other online outlets. Thanks, and we hope you enjoy reading it as much as we enjoyed writing it! PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk showing more than once on ps
Hello All , On Sat, 2 Jul 2005, Michael Stahl wrote: The system startup script /etc/init.d/asterisk calls the script /usr/sbin/safe_asterisk In safe_asterisk, the program is started with -c by default (console on TTY9). That explains why it is starting with a console, but not why it's running so many times! Here is what my system (FC3) shows: [EMAIL PROTECTED] sbin]# ps ax | grep asterisk 3371 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 3417 ?S 0:00 asterisk -vvvg -c 6846 ?S 0:00 asterisk -vvvg -c 6848 ?S 0:00 asterisk -vvvg -c 6849 ?S 0:00 asterisk -vvvg -c 6850 ?S 0:00 asterisk -vvvg -c 6853 ?S 0:01 asterisk -vvvg -c 6854 ?S 0:00 asterisk -vvvg -c 8479 pts/1S+ 0:00 grep asterisk Can anyone explain why asterisk is being launched 7 times? Thanks, OCG -Original Message- From: Luki [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk showing more than once on ps Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki Below ps is from a * server on slackware 10.0 using the command to start(**) . So I am not sure the '-c' is what is creating the multiple threads . linux-2.6 issue maybe ? Hth , JimL # ps -auxww | grep aster root 115 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 123 0.0 1.1 11916 5944 ?SJun30 0:01 /usr/sbin/asterisk -d -v -v -v root 125 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 130 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 131 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 132 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 139 0.2 1.1 11916 5944 ?SJun30 6:08 /usr/sbin/asterisk -d -v -v -v root 155 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 156 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 157 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 158 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v (**) # after expansion of variables . /usr/sbin/asterisk -d -v -v -v /var/log/asterisk/debug # sudo asterisk -V Asterisk CVS-HEAD-05/01/05-14:10:09 # uname -a Linux asterisk-1 2.6.11.8 #1 Sun May 1 12:04:14 MDT 2005 i686 unknown unknown GNU/Linu -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
Hello All , RFC = Request For Comments . STD = Standards Track Document(s) . Hth , JimL On Sat, 11 Jun 2005, Andrew Kohlsmith wrote: On Saturday 11 June 2005 11:35, Tracy Phillips wrote: That is *precisely* why the RFC is worded should -- it is optional. If the RFC said must then it is required. RFCs are worded very carefully as a general rule. I am just glad everyone doesn't have that attitude about RFCs. I'm not sure I understand -- I'm not making this up, RFCs use must and should very carefully. The latter is a guideline, and the former is a rule. I'm trying to find the link describing this but it's eluding me at the moment. I think this is a VERY good thing; RFCs are like the laws of the internet; they should not be open to interpretation since they define the protocols used to interoperate. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem
Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Autostart Asterisk on Slackware?
Hello Goran , Try this ... Please watch out for any wrapped
lines . Hth , JimL
cat EOF /etc/rc.d/rc.asterisk
#!/bin/sh
# --verbose
# Start the ASTERISK server.
PATH=/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin
NAME=asterisk
DESC=Asterisk PBX
# Full path to safe_asterisk script
SAFE_ASTERISK=/usr/sbin/safe_asterisk
ASTERISKDBN=asterisk
ASTERISKDN=/usr/sbin/${ASTERISKDBN}
ASTERISKCNFD=/etc/${ASTERISKDBN}
# Leave this set unless you know what you are doing.
export LD_ASSUME_KERNEL=2.4.1
# set -e
OPTS=-d -v -v -v
# usage rc.asterisk , start/stop/restart/reload
usage()
{
echo Usage: $0 {start|stop|restart|reload}
}
TCMD=$1
if [ -f ${ASTERISKDN} -a -d ${ASTERISKCNFD} ]; then
case $1 in
start) [ $TCMD = start ] \
echo -e \tStarting ${DESC}
if [ $OPTS = ]; then
$ASTERISKDN
else
$ASTERISKDN ${OPTS} /var/log/asterisk/debug 21
fi
;;
stop) [ $TCMD = stop ] \
echo -e \tStopping ${DESC}
$ASTERISKDN -rx 'stop now' 2/dev/null /dev/null
;;
reload) echo -e \tReloading ${DESC}
$ASTERISKDN -rx 'reload' 2/dev/null /dev/null
;;
restart)echo -e \tRestarting ${DESC}
$ASTERISKDN -rx 'restart gracefully' 2/dev/null /dev/null
;;
*) usage ;;
esac
else
echo -e \t${ASTERISKDN} or ${ASTERISKCNFD} , Does not exist .
echo -e \tPlease correct and re-reun this startup script
fi
EOF
On Tue, 15 Feb 2005, Goran Dj. wrote:
Maybe trivial question, but I cannot find an answer:
How to autostart Asterisk (daemon) on Slackware 10? I know that I should
put something in /etc/rc.d, but what?
--
+--+
| James W. Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux |
| [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP |
+--+
___
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Re: [Asterisk-Users] Cisco 7970 Firmware for the 7960G
Hello Michael , On Sat, 6 Nov 2004 [EMAIL PROTECTED] wrote: Hello, i´m thinking about buying one if the Cisco´s CP-7970G Phone. Does someone can confirm that it will work with asterisk? When last I checked on the 7970G Cisco was only providing SCCP protocol support . That was last June . I just rechecked their 7970G QA file it still shows only SCCP . I have not seen mention of SCCP support in * , but someone on this list sure knows . Hth , JimL I also have some trouble getting the newest firmware for my CP-7960G as Cisco doesn´t support people from outsite U.S. without a Support Contract(even with warranty) and it is very hard to get one here in Germany. Can someone please email me the latest upgrade for my two days old 7960G? :-) Regards Michael. -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.2
Hello Brian , Why wouldn't 'make clean' do just that ? Tia , JimL On Tue, 26 Oct 2004 [EMAIL PROTECTED] wrote: I did the trick, Wonderfull! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: dinsdag 26 oktober 2004 23:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk 1.0.2 rm all the .so's and try again. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, October 26, 2004 4:15 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk 1.0.2 Hello I compiled the new 1.02 over 1.01 My old asterisk 1.01 was compiled (on redhat 9.0) by downloading the src tarball from ftp.asterisk.com/pub/asterisk I did this the exact same way now, downloaded the 1.02 tarball, unpacked it, killed all asterisk 1.01 processes, issued a 'make' and 'make install', which seemed to compile without problems.. When starting the new version, asterisk exited with this error == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create ...snip... -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
