RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20
Re: [Asterisk-Users] Re: Advice on OS Choice
On Thu, 14 Oct 2004, David McNett wrote: On 14-Oct-2004, Kevin Walsh wrote: Red Hat have embedded their trademark all over their Enterprise editions so that they can restrict sales in that way. Red Hat still have an obligation to release the various GPLed components as usual but don't have to package the components nor create a downloadable CD ISO image. While this is correct, it is only half the story. The EULA on RHEL goes much further than relying on mere trademark protections. RedHat successfully uses their Trademark rights to prevent others from distributing RHEL but that has no sway over an existing user of RHEL. The EULA is where the real teeth are -- prohibiting even people who have purchased RHEL from using it in ways that RedHat prohibits. For example, it is not possible to purchase one copy of RHEL and install it on two machines. Nor are you allowed to run RHEL on a machine without having purchased support. I am unclear on how this is not a further restriction on the code (and therefore prohibited by the GPL) but the FSF appears unwilling to pursue the point. You cannot install a standard RHEL on a computer without copying non-gpl components. You could strip out all the non-gpl components and replace them. Then it would be legal to create a cpoy by installing on several computers. Which is what distributions such as WhiteBox and Tao have already done for you. Peter ___ 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] Re: Advice on OS Choice
On Thu, 14 Oct 2004, Joe Greco wrote: RedHat further encumbers RHEL with a EULA which extends the GPL and further restricts your rights to use the product. That, then, sounds like it might be a violation of the GPL. The GPL is, sadly, a maze of twisty little untested legal strategies, and even the IP lawyers don't know for sure. No, it is not. The restriction is placed on the non-gpl components. The gpl is very clear that shipping gpl and non-gpl components on the same media does not interfere with the gpl. Section 1 of the EULA says, essentially, go ahead, it's GPL. Section 2 of the EULA says, essentially, But we own our trademark and you cannot distribute that and we've stamped it all over the place. So if you distribute it you better damn well remove them all and woe to you if you fsck up. If this analysis is correct, this definitely flies in the /spirit/ of the GPL, which clearly does not expect people to have to modify files (and understand the side effects of the modifications) prior to redistributing them. The not spelled out part of this is that Red Hat itself is actually a trademark, and I suspect is stamped on copyright messages throughout the distribution, and /text has legally been considered an image/, so literal compliance with this EULA would require a redistributor to strip the Red Hat copyrights out of the files, and I expect that that would violate the GPL ... No, you do not. Attributions have no creative part, they are purely functional. Indeed, copyright messages are left intact in all the RHEL-based distributions. In fact, the non-gpl rpm:s are marked as such. There are some places where the argument may be used such as the naming of configuration files (/etc/redhat-release) and others. Those names are not purely functional (they are chosen at will and hence have a creative element). However, they are only distributed along with a gpl component. They themselfes are not under gpl. So this is ok too. Nothing to see here, move along folks. This is the same as it would have been without the eula. Creating copies requires a permission. For the gpl that is given but not for the non-gpl parts. In fact, without the eula you may not have been allowed to install the non-gpl parts of the distribution even if you bought a copy of RHEL. In some countries installation equals creating a copy. This is prohibited unless an eula grants that right. Peter ___ 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] SPA-2000's rebooting every hour or so...
How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL
[Asterisk-Users] Cisco firewalls and softphones
Hi, I know there has been some discussions regarding how you get a softphone to work across the Cisco PIX firewalls but I did not find any answer for the following scenario. Softphone X-Lite(Cisco VPN client) - Connects to PIX --- Asterisk - SIP client registered The problem is oneway voice (PIX problem). Well I had the same problem with Cisco´s IP softphone (H.323) and VPN client,this also hadone way voice ! I was lucky enough to have access toa Cisco engineer at the time and they also were unable to get it to work, they did give me some configs but still no luck. Somebody already mentioned they think this in fact might not be possible. So if that is the case then how are people connecting into their networks (securely) and using VoIP ? External SIP proxy and then ? I guess this is a multi faceted question but perhaps someone in Asterisk land could throw their opinions forward. Thank you for all the support. Matt Oulton. ___ 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
[Asterisk-Users] Newbie to Asterisk - VoIP end-to-end
Hi, After reading up on the Asterisk, I have a question: 1. Is there a software phone running on PC as a client that is compatible with Asterisk? My reason for asking is that I wonder if I can run voip end-to-end with Asterisk in between. Diagram: NetMeeting -- IP -- Asterisk -- (NetMeeting or its equivalent) or Cisco AS5xxx -- IP -- Asterisk -- (NetMeeting or its equivalent) Thanks, Kasey ___ 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] SPA-2000's rebooting every hour or so...
Hi Mine used to reboot on every call Clive On 15 Oct 2004 at 0:15, Mike Benoit wrote: How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06
Re: [Asterisk-Users] Am I stupid or is my card DOA.?
- Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 3:17 PM Subject: Re: [Asterisk-Users] Am I stupid or is my card DOA.? | | ** Niles Wrote | | | I had this exact same problem with my new TDM400P I just received | last | | week. | | It's very inconvenient not to be able to restart my phone server | | remotely, due to | | the TDM400P not restarting. | | I was able to successfully reboot my system earlier without | | cold-booting the server | | by unloading the zaptel modules before restarting the system. | | You will will to unload the modules in the opposite order that you | | loaded them, and may | | want to place them low in your /etc/rc.d/rc.K file (in slackware) for | | automation. | | (example for rc.K) | | | | # unload zaptel modules on shutdown. | | # asterisk should not be running when this occurs | | | | /sbin/modprobe -r wcfxs | | /sbin/modprobe -r xct1xxp | | /sbin/modprobe -r zaptel | | | | I haven't tested this extensively, but it has worked the one time I | | tested it. | | | | Niles | | | | | Niles | | I tried your suggestion, but still get the hang of the TDM card (4 fxo | modules) | | Regards | Greg | | | | Greg, | | did you receive any errors when attempting to unload the modules, and | was asterisk still running at that time? | Be sure asterisk is not running, and that you remove the modules in | reverse. | I only tried it once, so it could have been a red herring. | | Niles | Niles only error was the drivers were already unloaded doesn't matter if asterisk is running or not Greg ___ 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
[Asterisk-Users] zaptel compile error
Hi all, I am trying to compile Asterisk beginning with zaptel. Now I get 2 compile errors (see below). Can anyone give me a hint? Thanks Franz - sip:/usr/src/zaptel # make install cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.5-7.108' Makefile:438: .config: No such file or directory WARNING: Symbol version dump /usr/src/linux-2.6.5-7.108/Module.symvers is missing, modules will have CONFIG_MODVERSIONS disabled. CC [M] /usr/src/zaptel/zaptel.o /bin/sh: line 1: scripts/basic/fixdep: No such file or directory make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-7.108' make: *** [linux26] Error 2 sip:/usr/src/zaptel # ___ 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] Newbie to Asterisk - VoIP end-to-end
check out: http://www.voip-info.org/wiki-Asterisk+phones At 12:23 AM 10/15/2004 -0700, you wrote: Hi, After reading up on the Asterisk, I have a question: 1. Is there a software phone running on PC as a client that is compatible with Asterisk? My reason for asking is that I wonder if I can run voip end-to-end with Asterisk in between. Diagram: NetMeeting -- IP -- Asterisk -- (NetMeeting or its equivalent) or Cisco AS5xxx -- IP -- Asterisk -- (NetMeeting or its equivalent) Thanks, Kasey ___ 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 ___ 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] Looking for recommendations for a low-cost FXO toIP gateway.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Sent: 15 October 2004 02:41 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Looking for recommendations for a low-cost FXO toIP gateway. At first I thought the X100P was what I was looking for, but now it looks to me like the X100P does not have an IP interface, so it would require all audio to run through the CPU. I'm familiar with ATA186's, which I think are comparable to the IAXy box, and I'd just like to find something like that that provides an FXO interface. Can anyone help me? As far as I know there are quite few companies out there that do FXO Gateways. Mediatrix does a 2-port and 4-port anologue FXO model. The other solutions is a Sipura-3000. The Sipura 3000 will need it's own dialplan too, because it has an FXO, FXS and Eth port. So, you will have to do some configuration. Some people have reported different issues with it. one of them is over-heating. I haven't used any of the Mediatrix products, or any other FXO gateway, so I don't have personal experience. I hope it helps, Yiannis. ___ 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] Running Asterisk on Linksys Router
Hi, I don't know if I missed something on the recent posts regarding running * on the linksys boxes (couldn't make any sense of the gifs that were posted??)? Getting back to the original question, does anyone know where the firmware or source for a linksys box running * can be obtained? Aaron ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ 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] (Another) Queue log analyser
Hi there, Cheers for your suggestions, would be great to see the output of some other reports. Logins and logouts are available within the engine, just need to represent them in some way now. What do you suggest would be a good format? Typical duration of login? Only problem might be where someone hasn't logged out before their next login statement (no one here ever logs out, because they're all to slack :) Anything you can send me over would be much appreciated, I have no problems in giving you a pre-release copy so you can give some feedback too. Regards, Ben Merrills Griffin Internet T: 0870 8040862 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sheppard Sent: 14 October 2004 19:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (Another) Queue log analyser Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ 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 ___ 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 ___ 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
[Asterisk-Users] Prepaid authentication and accounting using Asterisk
Hi All, I wanted to know, if it's possible to use any available interface of Asterisk for authentication/authroization to plug some external billing application with it. The CSV file option is quite good for postpaid billing, but is there any way to do authentication of a PBX extension before allowing it to make a call and authorizing the caller a specific number of seconds during authorization. Thanks for your time, Regards, Jawad ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ 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
[Asterisk-Users] SIP - Asterisk - H323 Gateway
Hi, SIP - Asterisk - H323 Gateway What is the configuration of asterisk to make a call with this plan . My soft phone is registered to asterisk and i can establish a H323 channel from my asterisk to the Gateway. It's just about Dial function in extension.conf or other !! Thx ___ 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
[Asterisk-Users] FireFly w/ SIP
Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem ___ 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
[Asterisk-Users] CID troubles...
I have noticed that the data/time on the caller-id is incorrect on my Motorola 5ghz soho phone system. I go thru and reset the time on the handsets b/c there doesn't seem to be an option on the base for time. Then when I recieve a new call the caller-id is passed thru and the time gets screwed up again on *all* handsets. The configuration is incoming (PSTN/broadvoice) - * - SPA-3k. Does anyone know how the caller id parts work? Does it transfer the time with it? The time on the box is something like 20 mins off but the time that the handsets get reset to are 3-4hrs off when a caller-id is presented. Any pointers on where to go research would be great. Thanks, Jeff ___ 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: {SPAM?} [Asterisk-Users] Asterisk VIA SSH Tunnels
On Fri, 15 Oct 2004 11:12:30 +0900, Benjamin on Asterisk Mailing Lists wrote: On Thu, 14 Oct 2004 16:50:39 -0400, steve szmidt [EMAIL PROTECTED] wrote: Please don't use PPTP as a security solution, because it really isn't. It's so flawed you can even connect to it without having ANY encryption. Microsoft with their never ending wisdom have incorporated design flaws that make cryptographers and security professionals distrust it, and recommend against its use. Er, what's the news? Doesn't this apply to ANY product they make? Nonetheless, PPTP is widely deployed in corporate VPNs. You can allow unencrypted connections, but there is a setting to force encryption and even force encryption strength to 128 bits. I accept that it's not supremely secure. More advanced solutions are preferable. Buit it's included in every Windoze system and dead simple to setup. I think of it as about a secure as the WEP in my wirless lan. It discourages casual hacks of convenience, but that's about it. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 We're here for a good time, not a long time. So have a good time, the sun can't shine every day. - Trooper ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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
[Asterisk-Users] SNOM 190 Dial-Plan String Settings
I am having a problem with my new SNOM190 and my asterisk box. Incoming calls to the SNOM work perfectly, but when i dial-out I get a "Not Found: number dialed" on the SNOM display everytime I try, nothing shows up on the console of the asterisk box so its not even touching it. I have the latest 3.54 firmware on it and when I looked at the Line 1 setup for my asterisk box I released that in the SNOM phone there is nothing in my "Dial-Plan String" I take it it matches this inside the phone to choose which line to use in the SNOM phone. Unfortunately I am not finding much on the format of the Dial-Plan String in the SNOM phones. All I need is for it to send all calls regardless of format to the asterisk box. Anyone got any suggestions. James ___ 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] E911 support
I stand corrected, I forgot about CAMA trunks because we almost always just use PRI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Thursday, October 14, 2004 10:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E911 support There is a way called CAMA trunks, but that would not be supported in * to the best of my knowledge of *. Lyle - Original Message - From: Henry Devito [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, October 14, 2004 7:22 PM Subject: RE: [Asterisk-Users] E911 support AS Far as I have seen the only way to send E911 Info is over PRI. We have done the same thing with DID's as there is no way to send a different CSID over standard T1, at least in our area. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Thursday, October 14, 2004 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E911 support Here in Illinois, the requirement depends on the number of buildings and square footage of facility. But at one client, they had to do it. It was done via DID's. Each extension was assigned a DID number and we had to switch to a ISDN PRI circuit(they had a channelized T1 previously). The phone company then provides a method to map the DID number to the physical location information for the extension that placed the call. And that information is stored off site by the phone company in a database desigened exclusively for E991 use and not in the pbx. I was not involved in programing the PBX(wasn't an *), but was involved in with the vendor implementing this change. Lyle - Original Message - From: Steve Clark [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 14, 2004 1:50 PM Subject: [Asterisk-Users] E911 support Hi, A number of states now require business PBX's to supply location information to the station level. Does Asterisk provide this? Thanks, Steve -- They that give up essential liberty to obtain temporary safety, deserve neither liberty nor safety. (Ben Franklin) The course of history shows that as a government grows, liberty decreases. (Thomas Jefferson) ___ 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 ___ 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 ___ 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 ___ 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 ___ 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] MFC/R2 and Caller Id
I've an Asterisk connected to an Ericsson MD-110 PBX using MFC/R2. It's working fine but the caller id from the PBX to Asterisk is not set on the calls. From the debug i can see that the ANI is received but not set on the callerid field: Oct 7 19:38:37 WARNING[18451]: Offered on channel 0 (ANI: 7931, DNIS: 1931) On version 0.0.1c of Unicall the caller id issue is fixed. Bye, Leonardo -- Leonardo Gomes Figueira [EMAIL PROTECTED] ___ 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
[Asterisk-Users] Asterisk crashes on special Transfer with MGCP/ATA 186
Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one answers, then hangup, everything is fine, too. Any hints? mgcp debug on: -- Executing AGI(Zap/7-1, nuller.agi) in new stack -- Launched AGI Script /home/kpj/pbx/var/lib/asterisk/agi-bin/nuller.agi -- Accepting call from '01635571857' to '8551' on channel 0/1, span 3 -- AGI Script nuller.agi completed, returning 0 -- Executing Dial(Zap/7-1, MGCP/aaln/[EMAIL PROTECTED]||) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered Zap/7-1 gw-bzo*CLI mgcp debug on Usage: mgcp debug Enables dumping of MGCP packets for debugging purposes -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- Started music on hold, class 'default', on Zap/7-1 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '8' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '7' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- Stopped music on hold on Zap/7-1 Oct 15 13:32:58 NOTICE[100377]: chan_mgcp.c:1151 mgcp_fixup: mgcp_fixup(Zap/7-1, Zap/7-1MASQ) -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] Oct 15 13:32:58 WARNING[14350]: chan_mgcp.c:3033 handle_request: Transfer attempt failed ___ 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
[Asterisk-Users] Problem in DTMF Info message
i am sending this message to my asterisk first message: INFO sip:172.16.0.32 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.21 From: sip:[EMAIL PROTECTED] To: sip:172.16.0.32 Call-ID: [EMAIL PROTECTED] CSeq: 21 INFO Contact: sip:[EMAIL PROTECTED] Content-Type: application/dtmf-relay ContentLength: 26 Signal= 0 Duration= 160 it is not replying for all requests * only respons first time i am actualy trying for DTMF in Sip by INFO message can any one help me in this is this valid second message: INFO sip:172.16.0.32 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.21 From: sip:[EMAIL PROTECTED] To: sip:172.16.0.32 Call-ID: [EMAIL PROTECTED] CSeq: 21 INFO Contact: sip:[EMAIL PROTECTED] Content-Type: application/dtmf-relay ContentLength: 26 Signal= 1 Duration= 160 these two messages are correct ? ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ 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] Looking for recommendations for a low-cost FXO toIP gateway.
At first I thought the X100P was what I was looking for, but now it looks to me like the X100P does not have an IP interface, so it would require all audio to run through the CPU. I'm familiar with ATA186's, which I think are comparable to the IAXy box, and I'd just like to find something like that that provides an FXO interface. Can anyone help me? If you can use a box with 2 FXO and 2 FXS, take a look at the Planet VIP-450 http://www.planet.com.tw/product/product_dm.php?product_id=195menu_id=3 . We use the VIP-400 (H.323 version). It has lots of flexibility in the dial plan, IDs, etc. You can download the complete manual from the Planet site. Pros: Inexpensive, good voice quality, doesn't crash, excellent hardware reliability, good support for configuration problems. Cons: Many minor bugs and shortcomings, no support at all for getting these fixed, unless a big customer of Planet happens to experience the same trouble! --Stewart ___ 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
[Asterisk-Users] Re: CID troubles...
Figured it out...the SPA does a NTP call to the server running ntpd. As long as the timezone offset is up to date the caller-id will display the appropriate time and update all the handsets with the correct time. -Jeff ___ 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
[Asterisk-Users] Re: {SPAM?} Asterisk VIA SSH Tunnels
Tom Ivar Helbekkmo wrote: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes: And how many routers and firewalls out there do support OpenVPN? Do Cisco routers support it? Neither I, nor anyone else here, seems to be saying that OpenVPN is a replacement for IPsec. There's overlap, but there are applications that are more suited to one than to the other. As implementations of IPsec mature, its share should increase. (Today, you can still not take for granted that two IPsec VPN products will work seamlessly together.) I believe (but am more than ready to be proven wrong) that implementing the type of VPN that I'm using would be a real bitch with IPsec. I've got a portable computer that sends and receives quite a bit of sensitive data over insecure protocols, such as remote file system access -- and SIP, of course. :-) I carry this computer with me, and want to be able to use it wherever I can get hold of some sort of Internet connection. This might be by borrowing a real IP address somewhere, getting a DHCP-allocated RFC-1918 address behind some NAT gateway, or whatever. I have to expect there to be a firewall as well. An important requirement is that all sessions should survive when I suspend the computer, and then resume it somewhere else, where it gets a completely new access method to the Internet. For instance, while I'm directly connected by UTP cable at work, I open ssh sessions to various computers, I start a SIP-based soft phone, and, of course, I am connected to my remote file system server. I suspend the computer without logging out of anything, and later resume it in a place where there's a wireless hot spot that I'm allowed to access. I expect to be able to continue typing commands in those ssh sessions, receive telephone calls, and use the file system, immediately upon resuming. I need this to work completely NAT proof, and with no requirements for holes in firewalls other than being able to send a UDP packet out, and getting a responding packet back to the same port. It must also work without the suspend/resume: I need to be able to unplug my laptop's UTP cable to carry it into a meeting, and expect everything to keep working through a completely seamless transition to wireless mode. Of course, my laptop needs to have a fixed DNS name and IP address that never change, so it can be reached from the outside when needed. With OpenVPN running on my laptop, and on a VPN gateway system back home, this Just Works. OpenVPN handles the whole thing, it's well secured, all traffic is encrypted, and it automatically ensures that no traffic is sent or received by my laptop outside the VPN tunnel. I actually started looking into how to get comparable functionality based on IPsec, but my mind boggled, and now I do it the easy way. It works. I've done it. Tunnel 0.0.0.0/0 through IPSEC. Don't use AH. Make sure you're local networking is set up to use the extruded address that goes through your IPSEC tunnel. Of course, any firewall you come upon must allow allow UDP IDE and ESP packets through. But if they are intentionally blocking IPSEC, my guess is they're going to block all VPNs. ___ 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] Zap Channel wait for #
It has all gone very quiet - I still need this...I spent a fair bit of time looking at it but never got it to work. Needs someone with a bit more of an understanding of Asterisk's architecture really. Also, it should really go in app_dial so as to make it applicable across all channel types. Any volunteers? Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Jacob Sent: 15 October 2004 00:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zap Channel wait for # Hey all, I am using the c option when transferring to a zap channel to wait for a # before connecting the calling party. It works as advertised, however I would like to play a prompt to the called party. (ie Please press # to accept this call) http://bugs.digium.com/bug_view_page.php?bug_id=0002356 seems to be related. Anyone know what the status is? Or of there is a workaround of some sort? Thanks, Chris ___ 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 ___ 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] Dialogic D/300JCT-E1 support
On Fri, 2004-10-15 at 01:24 -0400, Donny Kavanagh wrote: Do the dialogic drivers from digium require those lame redhat 7.2/7.3 only drivers that intel released? Seeings as Digium just wrote a channel driver to connect the hardware driver to asterisk, I would guestimate that that would be correct to assume yes. More of the reason that not many people seem to be using the dialogic cards. -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: October 14, 2004 8:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dialogic D/300JCT-E1 support You would do well to ebay the card if you don't otherwise need it and then buy a Digium card. And you have to sign an NDA to get the drivers for a Dialogic card from Digium. bkw ___ 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 ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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
[Asterisk-Users] Re: how can I test canreinvite effectivness?
Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Try IPTRAF or TCPDUMP. Denis. Em Qui 14 Out 2004 19:12, Matthew Boehm escreveu: I'm not running X or any kind of GTK/GUI abilities on our asterisk server. I need some sort of ability to test wether sip canreinvite is working. If it is, then the network usage should be minimal/nonexistant because all voice packets should be going phone-to-phone. If it is not, then network usage would be high because all voice packets would be going phone-to-asterisk-to-phone Does anyone know of a nice ncurses or terminal based realtime network usage app? Or is there some other way in asterisk I can tell if the phones are talking to each other directly? This may be brute force and there may be more elegant methods, but I just monitor on the server with tethereal -R rtp and if I see packets then * is not releasing the media stream. The problem is that I have found that this can impair call quality if you leave it up, so I only do it to spot check. Also, I do an ethereal trace on the UA and look at the source/destination address of the rtp stream and that should tell you as well if the rtp is released. Thanks, Matthew ___ 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 ___ 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] zaptel compile error
On Fri, 2004-10-15 at 10:15 +0200, Franz Edler wrote: make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.5-7.108' Makefile:438: .config: No such file or directory Here's your hint. Did you make xconfig or menuconfig or something on your kernel? WARNING: Symbol version dump /usr/src/linux-2.6.5-7.108/Module.symvers is missing, modules will have CONFIG_MODVERSIONS disabled. Almost certainly not. -- Dave Cotton [EMAIL PROTECTED] ___ 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
[Asterisk-Users] Transmit re-INVITE before BYE is sent - why?
I have a question concerning re-INVITEs and how/why Asterisks sends them. On SIP to SIP calls with asterisk set up with canreinvite=yes, after a call is setup, media ip address:ports are renegotiated, 2 way rtp is established, and then one of the parties hangs up by sending a BYE, Asterisk goes through a re-INVITE process before forwarding on the BYE to the other party. I've shown the last part of the call flow (after the first re-INVITE to renegotiate the media stream) below where party A calls party B and after 2 way rtp has been established. Party AAsterisk Party B 5551200 5551212 ||| |-2 way RTP Peer to Peer-| ||| ||-BYE---| ||-200 OK| |--INVITE---|| |---TRYING--|| |---200 OK--|| |ACK|| |BYE|| |200 OK-|| In looking through the documentation, chan_sip.c file, associated web sites, configuration files, mailing lists, I have not been successful in finding out why it does this last re-INVITE but it appears that this may be performed as a clean up process allowing asterisk to possibly play a goodbye message or some other recording prior to hangup (rtp is transferred back to *). Is there something in the extensions.conf file that either turns on or turns off this activity. I have found that if you set canreinvite=no, this does not occur, but then * won't release rtp. My entry in extensions.conf is very simple: exten = _5551212,1,Dial(SIP/[EMAIL PROTECTED],,) exten = _5551212,2,Hangup ...and I've even tried taking out the Hangup step, but have not seen a difference. I was looking for a way to allow re-INIVITEs to renegotiate the rtp address:ports but not perform this last re-INVITE before a BYE. Any suggestions? Thanks. Tom Schroer ___ 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
[Asterisk-Users] Re: how can I test canreinvite effectivness?
Subject: Re: [Asterisk-Users] how can I test canreinvite effectivness? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Try IPTRAF or TCPDUMP. Denis. Em Qui 14 Out 2004 19:12, Matthew Boehm escreveu: I'm not running X or any kind of GTK/GUI abilities on our asterisk server. I need some sort of ability to test wether sip canreinvite is working. If it is, then the network usage should be minimal/nonexistant because all voice packets should be going phone-to-phone. If it is not, then network usage would be high because all voice packets would be going phone-to-asterisk-to-phone Does anyone know of a nice ncurses or terminal based realtime network usage app? Or is there some other way in asterisk I can tell if the phones are talking to each other directly? This may be brute force and there may be more elegant methods, but I just monitor on the server with tethereal -R rtp and if I see packets then * is not releasing the media stream. The problem is that I have found that this can impair call quality if you leave it up, so I only do it to spot check. Also, I do an ethereal trace on the UA and look at the source/destination address of the rtp stream and that should tell you as well if the rtp is released. Thanks, Matthew ___ 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 ___ 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
[Asterisk-Users] Invalid GSM data
I use my asterisk to SIP H323 Gateway. Softphone SIP - Asterisk - H323 Gateway - cellular phone I hear very well in my spftphone when i speak in my cellular But when i speak in my softphone the sound is very very very bad and i have this message in CLI console of asterisk : codec_gsm.c:164 gsmtolin_framein: Invalid GSM data What is the pb ??? ___ 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
[Asterisk-Users] Re: FireFly w/ SIP
I can tell you that you are not alone. It's an issue I believe with Firefly, and not in your configurations. Message: 8 Date: Fri, 15 Oct 2004 13:06:17 +0200 From: Willem de Groot [EMAIL PROTECTED] Subject: [Asterisk-Users] FireFly w/ SIP To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem ___ 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
[Asterisk-Users] Manager API and extension s
Hi all
When calls are set up using a macro, the extension in the status events coming from
action: status shows s. Does anybody know what to do to make the extension show
the correct value ?
My dialplan is like this:
[local]
exten = 8056,1,Macro(standardcall,SIP/t8)
exten = 8057,1,Macro(standardcall,SIP/t9)
[macro-standardcall]
exten = s,1,Dial(${ARG1},30,t)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
The status events are like this:
Event: Status
Channel: SIP/t8-549b
CallerID: User #7 8057
Account:
State: Up
Link: SIP/t9-72fe
Uniqueid: 1097738542.23
Event: Status
Channel: SIP/t9-72fe
CallerID: User #7 8057
Account:
State: Up
Context: macro-standardcall
Extension: s
Priority: 1
Seconds: 3
Link: SIP/t8-549b
Uniqueid: 1097738542.22
Thanks in advance
Martin
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RE: [Asterisk-Users] FireFly SIP Registration Interval
Awesome! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: Thursday, October 14, 2004 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FireFly SIP Registration Interval We'll add that to next version, should be out next week Deon Rodden wrote: I put FireFly on my mom's computer, but ran into a problem. She went home and was able to place calls from it (using her headset and such). But, she could not receive calls. I figured out the problem was with the registration, firefly doesn't re-register often enough, so the connection gets stale and the NAT Device forgets about the connection, so no new incoming calls can be made. I put X-Lite on her computer and changed the re-registration interval from the default of 3600 to 60 seconds. Now I can call her anytime. But, there's choppiness on the line. Her ability to transmit/upload/send voice to me is bad, I hear choppiness and such. FireFly worked fine, no choppiness, same router, same connection. I tried X-Lite and FireFly on my laptop but both perform equally. I like the simplicity and interface of firefly, it's nicer, anybody know of a way to change the sip registration interval? Anybody know of another program other than x-lite or firefly? One that doesn't have problems sending audio and one that allows you to change the sip registration interval? ___ 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 ___ 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] FireFly w/ SIP
I use FireFly w/ SIP all day long and it works great, except for the SIP registration interval which I was just told will be fixed in next weeks version. Are you using GSM or g711u? [remote-laptop] context=remoteusers type=friend username=remote-laptop secret=hiddenfromlist qualify=yes host=dynamic canreinvite=no dtmfmode=inband nat=yes callerid=John Doe 1235551212 accountcode=7499 amaflags=billing That's what I have in my sip.conf Then tell firefly to use the ip of your asterisk server as the Server. Give it the user id and password. Uncheck disable registration and check Active Always worked for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Willem de Groot Sent: Friday, October 15, 2004 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FireFly w/ SIP Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem ___ 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 ___ 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] Re: FireFly w/ SIP
FireFly is awesome, it's not giving quality issues like X-Lite is. FireFly's only problem was it wasn't registering with the server often enough, making that NAT box forget the connection and not allow incoming streams. Adam Hart said they would add it as an adjustable feature to the next version coming out next week. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leah Newmark Sent: Friday, October 15, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: FireFly w/ SIP I can tell you that you are not alone. It's an issue I believe with Firefly, and not in your configurations. Message: 8 Date: Fri, 15 Oct 2004 13:06:17 +0200 From: Willem de Groot [EMAIL PROTECTED] Subject: [Asterisk-Users] FireFly w/ SIP To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem ___ 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 ___ 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] zaptel compile error
On Fri, 2004-10-15 at 10:15, Franz Edler wrote: Hi all, I am trying to compile Asterisk beginning with zaptel. Now I get 2 compile errors (see below). Can anyone give me a hint? It would be nice if you did your homework before sending a msg to 8000+ people on this list. voip-info has all the info you need: http://www.voip-info.org/wiki-Linux+Fedora and read the instructions under the Fedora Core 2 heading. Always first check voip-info.org, the asterisk-user mailing list archives and google because most if not all newbie questions have already been answered. Regards, Patrick ___ 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
[Asterisk-Users] RE: Cisco firewalls and softphones (Matthew Oulton)
Speaking from personal experience using Cisco Callmanager and Cisco VPNs (not PIX, but Cisco VPNs hosted on routers with AIM cards), I can say that this is possible- but it's not easy. Essentially, the problem is not the VPN, it's NAT. In the cisco IP Softphone client, there's a rather disturbing section where you enter in your client's address- you can either have it pull the IP off the card, or set one permanently, or have it connect via HTTP to return the IP address. The important part is that the IP address chosen here must be the IP issued on the VPN, and *NOT* your current interface address. In other words, remove NAT entirely from the equation. Callmanager will accept the RTP stream from wherever it sees a valid connection- but, as we're all familiar with issues with NAT, and SIP, and H.323, Cisco Callmanager follows the standard and replies back to the IP that the client presents during call setup- hence, if the client presents a NATted address (from the callmanager's perspective), it will send the backhaul RTP to that address, and you get one-way audio. Some softphones are better at dealing with this than others. Long live IAX2! Paul Davidson ___ 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] too many ex-(boy|girl)friends
That's pretty good..
I have a similar situation, where I need to match all the area codes in
a particular state like:
exten = _[904|321|407|252]XXX,1,Dial..
But it doesn't work. I can get it to work with something along the lines of:
exten - _[904|321|407|352]X.,1,Dial
But I was hoping to be more specific.. other than specifying each area
code ala _904XXX,1,Dial. do you know of any way to do this?
Ben Wern
Maybe like this:
exten = s,5,DBGet(blacklisted=blacklisted/${CALLERIDNUM}) exten =
s,6,GotoIf(${blacklisted} = 1?hell|1)
You just have to put every blacklisted number in the Asterisk database as
it would be seen from the callerid number.
I this this solution is better than changing your extensions.conf every
time you change (boy|girl)friend.
Michel
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[Asterisk-Users] Always get 401 Unauthorized..that normal?
I always get a 401 Unauthorized result before the registration succeedes on these SIP phones. Is that normal? A REGISTER packet is sent, then a 100 Trying, then a 401 Unauthorized, then another REGISTER and another Trying, then OK. Is it normal to always get that 401? Why would registration be unauthorized then suddenly work? Or is this some algorithm that SIP uses to try different auth schemes? The phones are Cisco 7960 btw.. Thanks, Matthew ___ 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] rfc3389 support in chan_sip?
Throughout the discussion about this problem, I've learned more or less what the causes are. But. is rfc3389 support planned? thanks ___ 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] Always get 401 Unauthorized..that normal?
[EMAIL PROTECTED] wrote: Is it normal to always get that 401? Why would registration be unauthorized then suddenly work? Or is this some algorithm that SIP uses to try different auth schemes? Im see this too. I think the RFC says the UA shoudl try first without password, then with password. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Always get 401 Unauthorized..that normal?
Matthew Boehm wrote: I always get a 401 Unauthorized result before the registration succeedes on these SIP phones. Is that normal? A REGISTER packet is sent, then a 100 Trying, then a 401 Unauthorized, then another REGISTER and another Trying, then OK. I believe this is normal; most of the phones I've tested with initially attempt to register without specifying any authentication method. Asterisk then declines their registration, and they retry with authentication, which (presumably) succeeds. ___ 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] Always get 401 Unauthorized..that normal?
Yeah that is totally normal. To help prevent replay attacks the SIP device (Asterisk in this case) includes a authentication header in the Authentication Required response. This includes (among many other things) a random string that the initiator of the request (your phone) must include when creating the hash of its password. Hash sent = md5(password+random string) In short don't worry that's what is supposed to happen :-P Cheers alex -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: 15 October 2004 15:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Always get 401 Unauthorized..that normal? I always get a 401 Unauthorized result before the registration succeedes on these SIP phones. Is that normal? A REGISTER packet is sent, then a 100 Trying, then a 401 Unauthorized, then another REGISTER and another Trying, then OK. Is it normal to always get that 401? Why would registration be unauthorized then suddenly work? Or is this some algorithm that SIP uses to try different auth schemes? The phones are Cisco 7960 btw.. Thanks, Matthew Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ 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
[Asterisk-Users] Cisco to * problem
I am trying to connect a Cisco 3640 terminating a PRI to * with SIP. When I call into the PRI, the Cisco answers the call and sends it on to *, however there is no audio. The clue is, the following message out of *: Oct 15 07:50:58 NOTICE[1094289728]: chan_sip.c:2679 process_sdp: Content is 'multipart/mixed;boundary=uniqueBoundary', not 'application/sdp' Looking at the * code, this looks like a mismatch of some sort between * and Cisco, but I have tried every combination of codecs I can think if, and the problem doesn't change. Has anyone seen this message, or have a clue as to what it means? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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
[Asterisk-Users] app_queue manager API
Is there a way from the manager interface to obtain a listing of all the channels (callers) in a queue? I know as they join/part events are fired, but I'd like to obtain a listing of them when I connect to the manager interface. Any ideas how this can be done? Cheers, Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ 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] Cisco to * problem
I don't know anything about config of a Cisco 3640, do have a Cisco 5350 and have never seen it send SIP messages with multipart payloads. So can't really help you on that front. However I can tell you what that means. The INVITE request coming from the Cisco has multipart/mixed;boundary=uniqueBoundary as the type of payload in the message. Asterisk is expecting / requires application/sdp, in fact I would bet that most SIP endpoints would only support application/sdp. The payload in INVITEs (am simplifying a little) is where SIP devices detail what their media capabilities are, so that the other end point knows what types of audio / video it can send. Taking a complete guess I would suggest start by looking into audio codec settings on your Cisco 3640. Not much help I know but hopefully will give you a start. alex -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: 15 October 2004 15:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco to * problem I am trying to connect a Cisco 3640 terminating a PRI to * with SIP. When I call into the PRI, the Cisco answers the call and sends it on to *, however there is no audio. The clue is, the following message out of *: Oct 15 07:50:58 NOTICE[1094289728]: chan_sip.c:2679 process_sdp: Content is 'multipart/mixed;boundary=uniqueBoundary', not 'application/sdp' Looking at the * code, this looks like a mismatch of some sort between * and Cisco, but I have tried every combination of codecs I can think if, and the problem doesn't change. Has anyone seen this message, or have a clue as to what it means? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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 Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ [EMAIL PROTECTED] ___ 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] Re: Advice on OS Choice
Jason T. Nelson [EMAIL PROTECTED] wrote: In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said: If GNU/Linux was licensed under a BSD-style license then Red Hat could easily close the source - just as Apple did when they stole BSD code to create their OS/X effort. I don't believe that Red Hat would do that sort of thing anyway - those tactics are best left to Apple and Microsoft. Umm.. subtle but very important point here. Apple did not steal BSD code. BSD code cannot be stolen. It is given away as basically a gift. Stealing implies that the person you stole from has now lost something. Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Re: Advice on OS Choice
Jason T. Nelson [EMAIL PROTECTED] wrote: In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said: If GNU/Linux was licensed under a BSD-style license then Red Hat could easily close the source - just as Apple did when they stole BSD code to create their OS/X effort. I don't believe that Red Hat would do that sort of thing anyway - those tactics are best left to Apple and Microsoft. Umm.. subtle but very important point here. Apple did not steal BSD code. BSD code cannot be stolen. It is given away as basically a gift. Stealing implies that the person you stole from has now lost something. Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. That would be in violation of the BSD license. Maybe one of these days the GPL advocates will at least bother to read the license and get it right. By the way, assuming you've contributed code to Linux, did you get your check from RedHat for RHEL? Thought not. As usual, the irrational arguments like they weren't compensated are bandied about by GPL advocates, blissfully ignoring the fact that they wouldn't be compensated under the GPL, either. Please stop spreading inaccuracies and other FUD. If you can't at least speak with some mild accuracy about the differences between the two licenses, you are not competent to discuss the issue and should really not participate in such discussions. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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
[Asterisk-Users] RE: Cisco to * problem
See if you have the below configure under your dial peers or voice service voip. If you do, then issue this command no signaling forward unconditional signaling forward unconditional Kurt ___ 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] too many ex-(boy|girl)friends
How about:
exten = s,1,GotoIf($[${CALLERIDNUM} : ^(904|321|407|252)[0-9]{7}$] ?
2:3)
exten = s,2,Goto(somewhere,s,1)
exten = s,3,DoWhateverElse
On Oct 15, 2004, at 7:21 AM, Ben Wern wrote:
That's pretty good..
I have a similar situation, where I need to match all the area codes
in a particular state like:
exten = _[904|321|407|252]XXX,1,Dial..
But it doesn't work. I can get it to work with something along the
lines of:
exten - _[904|321|407|352]X.,1,Dial
But I was hoping to be more specific.. other than specifying each area
code ala _904XXX,1,Dial. do you know of any way to do this?
Ben Wern
Maybe like this:
exten = s,5,DBGet(blacklisted=blacklisted/${CALLERIDNUM}) exten =
s,6,GotoIf(${blacklisted} = 1?hell|1)
You just have to put every blacklisted number in the Asterisk
database as
it would be seen from the callerid number.
I this this solution is better than changing your extensions.conf
every
time you change (boy|girl)friend.
Michel
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Re: [Asterisk-Users] Invalid GSM data
You might have silence suppression turned on in the soft phone... turn it off. If that's not the culprit, use a different codec... maybe the soft phone just doesn't speak GSM right. On Oct 15, 2004, at 6:16 AM, CHAUVELIN Samuel wrote: I use my asterisk to SIP H323 Gateway. Softphone SIP - Asterisk - H323 Gateway - cellular phone I hear very well in my spftphone when i speak in my cellular But when i speak in my softphone the sound is very very very bad and i have this message in CLI console of asterisk : codec_gsm.c:164 gsmtolin_framein: Invalid GSM data What is the pb ??? ___ 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 ___ 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] Zap Channel wait for #
Does the option A(filename) not work for you? On Oct 15, 2004, at 5:38 AM, Robinson Tim-W10277 wrote: It has all gone very quiet - I still need this...I spent a fair bit of time looking at it but never got it to work. Needs someone with a bit more of an understanding of Asterisk's architecture really. Also, it should really go in app_dial so as to make it applicable across all channel types. Any volunteers? Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Jacob Sent: 15 October 2004 00:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Zap Channel wait for # Hey all, I am using the c option when transferring to a zap channel to wait for a # before connecting the calling party. It works as advertised, however I would like to play a prompt to the called party. (ie Please press # to accept this call) http://bugs.digium.com/bug_view_page.php?bug_id=0002356 seems to be related. Anyone know what the status is? Or of there is a workaround of some sort? Thanks, Chris ___ 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 ___ 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 ___ 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] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: The GPL protects the freedom of the source code and couldn't care less about the freedom of those who would seek to close the code. So, in other words, it's all right not to offer freedom to all. No, in other words freedom must be protected against those who would seek to deny it. Had you not licensed your software under the GPL, you could have benefitted from their efforts to extend your BSD-style copyrighted software. This is what has happened with companies like Apple and BSDi who have used the Berkeley UNIX codebase, as an example. Neither of those companies have contributed all of their changes back to the community, but then again, many of their changes would not be appropriate for distribution. That's not up to them to decide. Under the GPL, if you distribute modified code then you must publish your enhancements for the benefit of all. The team responsible for the core code can decide whether the contributed code is appropriate for distribution. The GPL basically says that if you don't want to distribute your changes, and want to use GPLed software, then start from scratch and write it all yourself. The BSD says I've spent a lot of time on this, but I'm happy for you to lock it up, make modifications and pretend that you wrote it all. People who say the GPL strips some of these freedoms really don't understand what freedom means. Yeah. GPL... let's slap some restrictions on what people can do. Surely encumbering software with restrictions on what you can do with it is more free than software that lets you do what you want. Isn't that an Ashcroft-esque definition of freedom? The whole point of the GPL is to protect the freedom of the code, for the benefit of all. If you consider the fact that you can't lock up the code and release it as a proprietary binary to be a restriction then I have no sympathy. Release your changes freely as open source and stop whining. In the remaining cases, you basically have people who don't want to contribute their changes back, for whatever reason (and there are valid reasons for this). a) This does not hurt a BSD licensed project, whereas b) The GPL'd project loses out if the person becomes motivated to go write a BSD licensed version of their product, so that they can then go and make their further undistributed changes in peace. This is especially damaging when there would have been a mix of noncontributed changes and also contributed changes coming back to the project, but instead now you have a competing project. That's all a nonsense. You started talking about people who don't want to contribute their changes back and then qualified it in (b) by saying that the project would have lost out. In this case, the project was in a no-win situation from the moment that person found it. With the GPL, if a person doesn't want to distribute the source and all changes then they can either (a) not distribute anything at all (b) create their own competing product. I welcome competition. You obviously have a proprietary outlook. c) The GPL'd project loses out if the person does something else entirely. If that person wasn't going to contribute then the project would have lost out regardless of its license. At least the GPL would have protected the project from an even worse situation - wholesale code theft and lock-up. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Re: Advice on OS Choice
On Fri, Oct 15, 2004 at 04:11:33PM +0100, Kevin Walsh wrote: Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. If the authors released their software under a BSD license then they INTENDED to allow this sort of use (advertising clause aside). I prefer the GPL too but there is nothing stupid about using a BSD style license. -- Ray ___ 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] Re: Advice on OS Choice
On Thu, 14 Oct 2004, Joe Greco wrote: RedHat further encumbers RHEL with a EULA which extends the GPL and further restricts your rights to use the product. That, then, sounds like it might be a violation of the GPL. The GPL is, sadly, a maze of twisty little untested legal strategies, and even the IP lawyers don't know for sure. No, it is not. The restriction is placed on the non-gpl components. The gpl is very clear that shipping gpl and non-gpl components on the same media does not interfere with the gpl. That doesn't seem to be the issue. The issue appears to be that they are shipping GPL components interspersed with items they have trademarked. Section 1 of the EULA says, essentially, go ahead, it's GPL. Section 2 of the EULA says, essentially, But we own our trademark and you cannot distribute that and we've stamped it all over the place. So if you distribute it you better damn well remove them all and woe to you if you fsck up. If this analysis is correct, this definitely flies in the /spirit/ of the GPL, which clearly does not expect people to have to modify files (and understand the side effects of the modifications) prior to redistributing them. The not spelled out part of this is that Red Hat itself is actually a trademark, and I suspect is stamped on copyright messages throughout the distribution, and /text has legally been considered an image/, so literal compliance with this EULA would require a redistributor to strip the Red Hat copyrights out of the files, and I expect that that would violate the GPL ... No, you do not. Attributions have no creative part, they are purely functional. Indeed, copyright messages are left intact in all the RHEL-based distributions. Correct, it would be a violation not to. However, I believe you've missed the point. This isn't about copyright. Red Hat appears to have said that you cannot distribute our trademark. There is a certain amount of law which allows a company to determine how its trademarked name (or other trademarks) are used, and I would be very wary of the situation where both copyright and trademark law applied, because I suspect the more restrictive would win out. If Red Hat distributes its logo image, under the GPL, but also has a notice on its website that the logo is a trademark with restrictions on use, you may not be violating the GPL by distributing it, but you may be breaking trademark rights. That's potentially actionable, and appears to be something the GPL didn't anticipate. Ecch. In fact, the non-gpl rpm:s are marked as such. There are some places where the argument may be used such as the naming of configuration files (/etc/redhat-release) and others. Those names are not purely functional (they are chosen at will and hence have a creative element). However, they are only distributed along with a gpl component. They themselfes are not under gpl. So this is ok too. Nothing to see here, move along folks. I'd check with a really good IPL and trademark lawyer before making that kind of a statement. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: By the way, assuming you've contributed code to Linux, did you get your check from RedHat for RHEL? Thought not. I was invited to take part in their IPO, under the friends of Red Hat scheme, which made me over £120,000 profit on my investment. Does that count? Thought not. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] SPA-2000's rebooting every hour or so...
I'm using the ulaw codecs, and checking again, I just realized we have one SPA-3000 in the mix behaving exactly like the SPA-2000's. Changing the registration expire time to 6hrs didn't seem to make any noticeable difference unfortunately. On Fri, 2004-10-15 at 09:42 +0200, [EMAIL PROTECTED] wrote: Hi Mine used to reboot on every call Clive On 15 Oct 2004 at 0:15, Mike Benoit wrote: How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL
[Asterisk-Users] Prepaid vs. Prepaid modified
Hi all, Anyone know what the differences are between the Prepaid and the Prepaid-modified apps is? The provided docs don't say much. David Filion ___ 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] Zap Channel wait for #
No - this plays the message AFTER the # is pressed, not before -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Scott Sent: 15 October 2004 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap Channel wait for # Does the option A(filename) not work for you? On Oct 15, 2004, at 5:38 AM, Robinson Tim-W10277 wrote: ___ 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
[Asterisk-Users] New Project - IP Phone Sources
Hello, Background: Old to UNIX Linus, New to list. A techie Dad that supports local k-8 school that my kids go to. More background: Recently the school wanted to put phones in all the classrooms for teacher communications to/from the office. Another Dad in the telecom business spec'ed out a standard PBX with wiring, etc. Needless to say it was Expensive with a Captitol E. Anyway I started looking around at open source and found Asterisk. We currently have a complete switched network within the school (jsut replaced all hubs with switches) and have multiple PC's in each classroom as well as the front office. We also run RH Linux for our webserver, email server, file server, Websense server, and library software server. Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. Many thanks in advance. BTW, the school is: www.sainttheresaschool.org stew Stewart M. Ives SofTEC USA WebSite: www.softecusa.com ___ 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] Re: Advice on OS Choice
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said: Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. The BSD license says nothing about compensation; have you read it lately? * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright *notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright *notice, this list of conditions and the following disclaimer in the *documentation and/or other materials provided with the distribution. Nope, not in there. Sorry. You are mocking software authors who make a CONSCIOUS choice to use the BSD license (or other BSD-style licenses) by calling it stupid. Perhaps they are uncomfortable with some of the restrictions imposed by the GPL? -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. pgpDKA2WtCNcn.pgp Description: PGP signature ___ 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] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: The GPL protects the freedom of the source code and couldn't care less about the freedom of those who would seek to close the code. So, in other words, it's all right not to offer freedom to all. No, in other words freedom must be protected against those who would seek to deny it. Had you not licensed your software under the GPL, you could have benefitted from their efforts to extend your BSD-style copyrighted software. This is what has happened with companies like Apple and BSDi who have used the Berkeley UNIX codebase, as an example. Neither of those companies have contributed all of their changes back to the community, but then again, many of their changes would not be appropriate for distribution. That's not up to them to decide. Under the GPL, if you distribute modified code then you must publish your enhancements for the benefit of all. The team responsible for the core code can decide whether the contributed code is appropriate for distribution. Yes, and that's fine, but it's not free. You've encumbered the software with restrictions. The GPL basically says that if you don't want to distribute your changes, and want to use GPLed software, then start from scratch and write it all yourself. The BSD says I've spent a lot of time on this, but I'm happy for you to lock it up, make modifications and pretend that you wrote it all. Pretend that you wrote it all? No, big license violation. Go read the BSD license and don't even think of replying to this message until you do. People who say the GPL strips some of these freedoms really don't understand what freedom means. Yeah. GPL... let's slap some restrictions on what people can do. Surely encumbering software with restrictions on what you can do with it is more free than software that lets you do what you want. Isn't that an Ashcroft-esque definition of freedom? The whole point of the GPL is to protect the freedom of the code, The freedom of the code? People have freedoms. Code is an object. That's like saying to protect the freedom of my car or to protect the freedom of my gun or to protect the freedom of my computer. Only a bunch of computer geeks who have read too many rights of robots scifi stories would realistically believe that somehow code could have any rights. How about the right to exist? Will it become illegal for me to delete software from a computing system? If code doesn't even have that basic right, then how is it you're arguing for rights so much more abstract? Come on, get real. for the benefit of all. If you consider the fact that you can't lock up the code and release it as a proprietary binary to be a restriction then I have no sympathy. Release your changes freely as open source and stop whining. Sometimes the changes are not appropriate to release as open source. Sometimes you /can't/, for legal or liability reasons. In the remaining cases, you basically have people who don't want to contribute their changes back, for whatever reason (and there are valid reasons for this). a) This does not hurt a BSD licensed project, whereas b) The GPL'd project loses out if the person becomes motivated to go write a BSD licensed version of their product, so that they can then go and make their further undistributed changes in peace. This is especially damaging when there would have been a mix of noncontributed changes and also contributed changes coming back to the project, but instead now you have a competing project. That's all a nonsense. You started talking about people who don't want to contribute their changes back and then qualified it in (b) by saying that the project would have lost out. In this case, the project was in a no-win situation from the moment that person found it. No. Look at the case of Apple, as one trivial counterexample. With the GPL, if a person doesn't want to distribute the source and all changes then they can either (a) not distribute anything at all (b) create their own competing product. I welcome competition. You obviously have a proprietary outlook. No, I have a practical outlook, tempered by years of experience as a software author, including in fields such as medical monitoring. Have you ever written code for something like a medical monitor? For numerous reasons, you don't want that code available to the public. You don't need some not-smart-enough hospital techie trying to make changes to it, figuring out how to override the safeguards and then installing it on your equipment, and then suddenly having liability issues. That doesn't mean that during the course of coding that project, that you run across a nice high performance GPL'd line drawing algorithm, which is perfect except that it doesn't draw antialiased lines, and while you would have no problem writing and returning the antialiased
RE: [Asterisk-Users] New Project - IP Phone Sources
You have more options than you know. You could go with a channel bank if you want to keep support for the analog phones in the classrooms now(my school had them) or you could goto the next step with the sip phones. I have looked around and found a couple vendors to be fairly inexpensive. Check this link out: http://www.voip-info.org/wiki-VOIP+Phones Check under hardphones. It's a very good resource for the information your looking for. As far as the dialplan. It would take no time to build what your looking for and get everything setup. Got any questions feel free to drop me a email .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stewart M. Ives Sent: Friday, October 15, 2004 12:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Project - IP Phone Sources Hello, Background: Old to UNIX Linus, New to list. A techie Dad that supports local k-8 school that my kids go to. More background: Recently the school wanted to put phones in all the classrooms for teacher communications to/from the office. Another Dad in the telecom business spec'ed out a standard PBX with wiring, etc. Needless to say it was Expensive with a Captitol E. Anyway I started looking around at open source and found Asterisk. We currently have a complete switched network within the school (jsut replaced all hubs with switches) and have multiple PC's in each classroom as well as the front office. We also run RH Linux for our webserver, email server, file server, Websense server, and library software server. Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. Many thanks in advance. BTW, the school is: www.sainttheresaschool.org stew Stewart M. Ives SofTEC USA WebSite: www.softecusa.com ___ 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 ___ 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] New Project - IP Phone Sources
See comments inline... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stewart M. Ives Sent: Friday, October 15, 2004 12:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Project - IP Phone Sources Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Stew, Asterisk is definitely the perfect application for which you are trying to accomplish. You could even integrate asterisk into the current PBX if you wanted. But simply putting up an asterisk server and some sort of IP hardphone would work perfect for your scenario. Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. The most basic phones, I think many will agree, are Grandstreams. From what I have read they seem to have pretty good integration with *. I have never used these, but have used Polycom IP 500's. For a business, in my case a law firm, these phones have worked pretty reliably. Best regards, - Brent ___ 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] New Project - IP Phone Sources
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stewart M. Ives Sent: 15 October 2004 17:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Project - IP Phone Sources Hello, Background: Old to UNIX Linus, New to list. A techie Dad that supports local k-8 school that my kids go to. More background: Recently the school wanted to put phones in all the classrooms for teacher communications to/from the office. Another Dad in the telecom business spec'ed out a standard PBX with wiring, etc. Needless to say it was Expensive with a Captitol E. Anyway I started looking around at open source and found Asterisk. We currently have a complete switched network within the school (jsut replaced all hubs with switches) and have multiple PC's in each classroom as well as the front office. We also run RH Linux for our webserver, email server, file server, Websense server, and library software server. Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? -Correct! Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. -I don't think you can get any less basic than the Grandstream Budgetone 101. The do still have features though. Yiannis. ___ 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] New Project - IP Phone Sources
Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Yes. You will spend some time configuring stuff, but it should work just fine. Between the list and #asterisk on irc.freenode.net you shouldn't have too much trouble. Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. Check out: http://www.voip-info.org/wiki-VOIP+Phones I think the Grandstream BudgeTones are the cheapest ones you'll find. You could also use an adapter to use existing analog phones, but I don't think that'll save much money. -Michael ___ 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] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: By the way, assuming you've contributed code to Linux, did you get your check from RedHat for RHEL? Thought not. I was invited to take part in their IPO, under the friends of Red Hat scheme, which made me over £120,000 profit on my investment. Does that count? Thought not. So they let you buy some stock. Did you have to pay for it? Did they give stock to all Linux contributors whose work they benefitted from? See, I bought stock, years ago, while working for a company. I don't consider the return on investment to be compensation for having worked there. That'd be kinda silly. If they gave the stock to you, and to all the other contributors whose work they benefitted from, then I am very much mistaken, and I apologize. If they didn't, then I don't think you got your check from RedHat for RHEL, and that brings up the question do you approve of what they did with your code? Specifically, did you expect that they would sell your code in a distribution that essentially forbids redistribution? Hopefully we can at least agree that what they did was distasteful at best. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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
[Asterisk-Users] Cannot reach a SIP device
I am trying to call a my friend who has GS HandyTone-486 behind a firewall but it goes to his voicemail straightway. Surprisingly, he can call me fine. I also see that his device is properly registered. Can anyone help me resolve this problem. In my sip.conf I do have canreinvite=no and nat=yes. In the GS HandyTone, he has set use random port = yes and NAT traversal = yes When he calls me, there is no problem at all, audio is fine too. Thanks, -- sudhir Here are some debug messages from the Server: cequip2*CLI database show .. /SIP/Registry/3110 : 168.243.154.92:63210:300:3110 . cequip2*CLI sip debug ip 168.243.154.92 SIP Debugging Enabled for IP: 168.243.154.92 After I call him from my extension: Peer RTP is at port 192.168.2.4:0 -- Executing Dial(SIP/4390-8620, SIP/3110|15|rt) in new stack We're at 66.251.6.188 port 10502 12 headers, 7 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:63210 SIP/2.0 Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8 From: Sudhir Kumar sip:[EMAIL PROTECTED];tag=as009f251d To: sip:[EMAIL PROTECTED]:63210 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 15 Oct 2004 16:50:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 136 v=0 o=root 26933 26933 IN IP4 66.251.6.188 s=session c=IN IP4 66.251.6.188 t=0 0 m=audio 10502 RTP/AVP a=silenceSupp:off - - - - (NAT) to 168.243.154.92:63210 -- Called 3110 cequip2*CLI Sip read: SIP/2.0 415 Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8 From: Sudhir Kumar sip:[EMAIL PROTECTED];tag=as009f251d To: sip:[EMAIL PROTECTED]:63210;tag=9a40e4e5aafd25aa Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream HT486 1.0.5.10 Content-Length: 0 8 headers, 0 lines -- Got SIP response 415 back from 168.243.154.92 Transmitting: ACK sip:[EMAIL PROTECTED]:63210 SIP/2.0 Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8 From: Sudhir Kumar sip:[EMAIL PROTECTED];tag=as009f251d To: sip:[EMAIL PROTECTED]:63210;tag=9a40e4e5aafd25aa Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 168.243.154.92:63210 == No one is available to answer at this time -- Executing VoiceMail(SIP/4390-8620, 3110) in new stack -- Playing 'vm-intro' (language 'en') Destroying call '[EMAIL PROTECTED]' == Spawn extension (default, 3110, 2) exited non-zero on 'SIP/4390-8620' ___ 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] Zap Channel wait for #
Message: 6 Date: Fri, 15 Oct 2004 08:29:01 -0700 From: Chad Scott [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap Channel wait for # To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed Does the option A(filename) not work for you? The A(filename) option will not fire until after the called party presses #. When they first answer the call they only get silence... I have a find me routine that I use but I forward the call with the original caller ID. There are two reasons why I want to use this option... One is to let the called party know that this call is coming via asterisk and the other is to prevent voicemail from grabbing the call before I have a chance to pull it back and move on to the next step. ~c ___ 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
[Asterisk-Users] Re: Advice on OS Choice
In article [EMAIL PROTECTED], Kevin Walsh [EMAIL PROTECTED] wrote: That's not up to them to decide. Under the GPL, if you distribute modified code then you must publish your enhancements for the benefit of all. The team responsible for the core code can decide whether the contributed code is appropriate for distribution. That's not how I understand the GPL. My understanding is that the GPL gives me the freedom to take some GPLed code, modify it, and distribute the modified code to whomsoever I choose, for free or for a payment. I must also make the source code freely available (on request, if I prefer) to anyone to whom I distribute binaries. It does *not* compel me to distribute my version to everyone, but I also cannot prevent those people I give or sell it to from passing it on in binary and/or source form to anyone else if they choose to. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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
[Asterisk-Users] New Open Source Project: Asterisk Management Portal
Salutations, In hopes of accelerating the adoption of Asterisk and changing the landscape of the small business marketplace, we are contributing our administration interface to a new project that aims to bundle best-of-breed applications to produce a canned (but fully functional) turnkey small business phone system. Details of the project can be found here: http://amp.voxbox.ca Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ 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] app_queue manager API
Ben Check out Action: QueueStatus - it'll list the stats for each queue as well as listing each queue member verbosely. Cheers Paul ___ 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] New Project - IP Phone Sources
Hi Stewart, Nice project! Something I'd certainly love to be doing myself. Anyway, the following replies I've made to your questions are based on my experience and past research. There may be better/cheaper alternatives. In any case, I hope it helps: On Fri, 2004-10-15 at 12:05 -0400, Stewart M. Ives wrote: Hello, [snip] Question: If I just want to provide IP Telephony within the school and have no outside connections to the local phone system I suspect I can install Asterisk on a RH Linux server and plug in a bunch of IP Telephones on the network, config it all and it will work. The only cost to the school would be the IP Telephones. Correct?? I know it would involve a bit more configuration and planning as I have stated but basically is the idea correct?? Question: What phones or types of phones should I be looking at. I suspect there are new ones coming out every day. I'm just interested in the most basic phone to plug into the network. Nothing fancy, basic, basic, basic. I also know I can use soft phones but do not want to go there as it makes just another application we have to be responsible for on the desktop. Many thanks in advance. Pretty much. You have the following options as far as I can see (and I'm sure there's more): 1) FXS Adapter - The IAXy[1] is a nice (and cute) device which allows you to connect a single analog telephone and provide VoIP connectivity using IAX to your Asterisk server. Buying the device helps support Asterisk. The only catch is that it only supports one analog phone. Keeping price in consideration, the only other device I would recommend is the Sipura SPA-2000 which supports 2 analog telephones per device (you would need one SPA-2000 per 2 classrooms (one analog phone per classroom)) 2) Digium TDM40B[2] (includes the TDM400P card plus the 4 FXS modules): This configuration provides 4 x FXS (analog telephone) ports on a single half-length PCI card. I just checked the Digium site and they're selling the TDM40B for $305 (works out to be around $76 per telephone). Certainly the best way of doing it, IMHO. Keep in mind with this solution you would need telephone wiring FROM the Asterisk server where the TDM40B lives to all the classrooms. With the IAXy or the SPA-2000 you just need telephone wiring from the unit itself to each classroom it's providing VoIP to. Great thing about this solution is that you can mix and match. If, for instance, the school decided to get a telephone line hooked up to the system, you can buy a FXO module and swap it for an unused FXS module, or configure it however you want. 3) VoIP Telephones: Cheapest is the infamous Grandstream[3] BudgeTone (AKA BarbieTone). Well, actually, I shouldn't say infamous since I've not had a problem with them myself, but you'll find many reports from other users on the mailing list archives about the myriad of problems you can have with them. If you already have a network connection going into each classroom, this (or the FXS adapters) may be the best option. Hope this helps! Best regards, Gonzalo [1] http://www.digium.com/index.php?menu=iaxy [2] http://www.digium.com/index.php?menu=wildcard_tdm400p2 [3] http://www.grandstream.com/y-bt100.htm ___ 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] app_queue manager API
Ah cheers, It seems to have changed to add the event ' QueueEntry' from when I last looked at the src. Cheers for your help Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: 15 October 2004 18:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] app_queue manager API Ben Check out Action: QueueStatus - it'll list the stats for each queue as well as listing each queue member verbosely. Cheers Paul ___ 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 ___ 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] New Open Source Project: Asterisk Management Portal
On Fri, 2004-10-15 at 18:56, Jason Becker wrote: Salutations, In hopes of accelerating the adoption of Asterisk and changing the landscape of the small business marketplace, we are contributing our administration interface to a new project that aims to bundle best-of-breed applications to produce a canned (but fully functional) turnkey small business phone system. Details of the project can be found here: http://amp.voxbox.ca Hi Jason, Thank you very much for your contribution. Small remark: no website is complete withoutscreenshots! :) Regards, Patrick ___ 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] Re: Advice on OS Choice
Tony Mountifield [EMAIL PROTECTED] wrote: Kevin Walsh [EMAIL PROTECTED] wrote: That's not up to them to decide. Under the GPL, if you distribute modified code then you must publish your enhancements for the benefit of all. The team responsible for the core code can decide whether the contributed code is appropriate for distribution. That's not how I understand the GPL. My understanding is that the GPL gives me the freedom to take some GPLed code, modify it, and distribute the modified code to whomsoever I choose, for free or for a payment. I must also make the source code freely available (on request, if I prefer) to anyone to whom I distribute binaries. It does *not* compel me to distribute my version to everyone, but I also cannot prevent those people I give or sell it to from passing it on in binary and/or source form to anyone else if they choose to. Ok. You're not obliged to submit your modifications back to the core project, although it is polite to attempt to do so. You are obliged to provide (or make available) the full source to both the original project and your modifications, to whomever you distribute your version. Your modifications might make it back into the official version, or they might not - that decision would be up to the core team to decide. Your understanding of the GPL appears to be correct, and I'm glad to see that it doesn't contradict my understanding. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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
[Asterisk-Users] RE: Cannot reach a SIP device (Sudhir Kumar)
Never mind, I found out what the problem was. On investigating the response 415, I discovered that codecs could not be negotiated properly. I changed the codecs on server and HandyTone, works great now. -- sudhir ___ 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] New Open Source Project: Asterisk Management Portal
Patrick wrote: Thank you very much for your contribution. Small remark: no website is complete withoutscreenshots! :) Regards, Patrick Hi Patrick, Right you are! We'll work on getting some up. In the mean time, have a look at: http://www.voxbox.ca/products.php?display=4 The interface is exactly the same, minus the 'voxbox' brand. Cheers -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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] Re: Advice on OS Choice
In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said: Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. The BSD license says nothing about compensation; have you read it lately? * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright *notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright *notice, this list of conditions and the following disclaimer in the *documentation and/or other materials provided with the distribution. Nope, not in there. Sorry. Perhaps you should have read my article before rushing to respond. If you did then you'd notice the as allowed by the stupid BSD license part. You are mocking software authors who make a CONSCIOUS choice to use the BSD license (or other BSD-style licenses) by calling it stupid. Perhaps they are uncomfortable with some of the restrictions imposed by the GPL? The original author would have no restrictions imposed upon them. The restriction applies to people who would like to close the source and incorporate it into a preparatory, closed source, product. Perhaps you'd be happy to see Asterisk released under a BSD license, rather than the GPL. Perhaps you'd also be happy to see a Microsoft PBX with embrace and extend features and a future defacto standard closed-source MS-IAX protocol. The GPL prevents this and thereby protects software freedom, the BSD license would not. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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
[Asterisk-Users] FXS port to use an Analog phone as a door phone.
Hello all. I have a problem which I do not find a solution to. I need to have a Plain jane analog phone when you pick it up With you dialing any numbers (Dial Pad is broken) it dials automatically for you. This is going to be for a door phone. Or in another case it's for a phone in an elevator. Remember there not plugged into an FXO but normal FXS ports. I have put the zap port context like this. But it does not work. [door] exten = s,1,Answer exten = s,2,Dial(Zap/2) exten = s,3,Hangup In the context for Zapata.conf signaling=fxo_ks Context=door immediate=yes Channel=1 Does anyone have any idea? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721x121 Fx: 305-574-0212 ___ 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] New Open Source Project: Asterisk Management Portal
BRILLIANT MOVE! Kudos to you for your decision to contribute your efforts back into the community! Regards, Jim Van Meggelen Core Telecom Group [EMAIL PROTECTED] 416-429-1304 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker Sent: October 15, 2004 12:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Open Source Project: Asterisk Management Portal Salutations, In hopes of accelerating the adoption of Asterisk and changing the landscape of the small business marketplace, we are contributing our administration interface to a new project that aims to bundle best-of-breed applications to produce a canned (but fully functional) turnkey small business phone system. Details of the project can be found here: http://amp.voxbox.ca Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.775 / Virus Database: 522 - Release Date: 08/10/2004 ___ 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] Re: Advice on OS Choice
Joe Greco [EMAIL PROTECTED] wrote: Have you ever written code for something like a medical monitor? For numerous reasons, you don't want that code available to the public. You don't need some not-smart-enough hospital techie trying to make changes to it, figuring out how to override the safeguards and then installing it on your equipment, and then suddenly having liability issues. Making the code available and allowing unqualified people to tinker with live medical equipment are two separate issues. You're getting confused now. That doesn't mean that during the course of coding that project, that you run across a nice high performance GPL'd line drawing algorithm, which is perfect except that it doesn't draw antialiased lines, and while you would have no problem writing and returning the antialiased line code back to that project, you don't want your entire product becoming subject to the GPL. If you don't like the terms of the license chosen by the author(s) of another project then write your own code. If you want to take some GPLed code and don't want to release your project as open source, under the GPL, then write your own code. I don't see the problem. That's (close to) real world. In reality, we had a somewhat larger example (plus some other miscellaneous examples) of something that would have been nice to use, and which would have benefitted from returned changes, had they not been licensed under GPL. We did, in fact, make great use of X11, contributed various code fixes and other things back to that project, though the driver I wrote for the propietary touchscreen stuff was not sent back to MIT... what would the point have been? If you haven't realised the point of open source software and software freedom by now then I can't really see the benefit in explaining it to you again. Perhaps you should apply for a job at Microsoft or Apple. At least the GPL would have protected the project from an even worse situation - wholesale code theft and lock-up. Theft? Lock-up? No. That's what happens when someone actually breaks a license. Exactly. The BSD would allow this sort of thing to continue legally. The GPL would not, and purposefully prevents open source software from being closed. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] New Open Source Project: Asterisk Management Portal
Salutations, In hopes of accelerating the adoption of Asterisk and changing the landscape of the small business marketplace, we are contributing our administration interface to a new project that aims to bundle best-of-breed applications to produce a canned (but fully functional) turnkey small business phone system. Details of the project can be found here: http://amp.voxbox.ca After looking at the screenshots, I must say it looks very promising. Great work, and thank you for contributing back to the community! Regards, Gonzalo ___ 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
[Asterisk-Users] grandstream bt-486 can only dial with #
I have a grandstream BT-486 in the lab running 1.0.5.11 firmware. For the past three days I've had no trouble dialing out without hitting #. I had the setting for using # as dial key to no in the config. Today the BT wouldn't pass outgoing calls. I turned on # as dial key and it works now if I hit # at the end. I have changed nothing on the BT-486 and nothing on the * box it is connected to. Anybody seen this happen before? ___ 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
[Asterisk-Users] CHANUNAVAIL = CHANUNAVAIL doesn't eval properly
Here is the relevant dialplan:
exten = _3XXX,1,Dial(SIP/${EXTEN},15,tr)
exten = _3XXX,2,Voicemail([EMAIL PROTECTED])
exten = _3XXX,102,GotoIf($[${DIALSTATUS}==CHANUNAVAIL]?i,1:103)
exten = _3XXX,103,Voicemail([EMAIL PROTECTED])
exten = i,1,Playback(invalid)
exten = i,2,Hangup()
What 'should' happen is that if someone dials any extension starting with 3,
Dial attempts to dial it. If there is no such channel, set DIALSTATUS and
goto priority n + 101. Then check result of DIALSTATUS. If DIALSTATUS is
equal to CHANUNAVAIL then goto extension 'i' else goto priority 103.
Here is an attempt to dial non-existant extension 3652:
Oct 15 13:10:06 WARNING[1116730816]: chan_sip.c:1384 create_addr: No such
host: 3652
Oct 15 13:10:06 NOTICE[1116730816]: app_dial.c:742 dial_exec: Unable to
create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing GotoIf(SIP/3044-9a9c,
CHANUNAVAIL=CHANUNAVAIL?cytel-internal|i|1:103) in new stack
-- Goto (cytel-internal,3652,103)
-- Executing VoiceMail(SIP/3044-9a9c, [EMAIL PROTECTED]) in new stack
Oct 15 13:10:06 WARNING[1116730816]: app_voicemail.c:1367 leave_voicemail:
No entry in voicemail config file for '3652'
It seems that CHANUNAVAIL does not equal CHANUNAVAIL in this case. This is
incorrect. Any ideas?
Thanks,
Matthew
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RE: [Asterisk-Users] Re: Advice on OS Choice
Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. The BSD license says nothing about compensation; have you read it lately? Come on guys, can we get back to * and stop this. 99% of the folks on this list don't care about license perceptions and personal opinions. Take it off list. ___ 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] FXS port to use an Analog phone as a door phone.
On Fri, 2004-10-15 at 13:27, Ariel's Hotmail wrote: Hello all. I have a problem which I do not find a solution to. I need to have a Plain jane analog phone when you pick it up With you dialing any numbers (Dial Pad is broken) it dials automatically for you. This is going to be for a door phone. Or in another case it's for a phone in an elevator. Remember there not plugged into an FXO but normal FXS ports. You can get phones that will dial a # when they go off-hook. Try http://www.redhotphones.com or their other websites listed on their site. Some are pure doorphones (black box with speaker and a call button) and some are phones without a dial. ___ 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
[Asterisk-Users] Sample advanced call routing standard extension
[default]
;
; Eric Wieling
;
exten = 2120,1,SetVar(DND=)
exten = 2120,2,SetVar(CFU_DEST=)
exten = 2120,3,SetVar(CFU_TIMEOUT=)
exten = 2120,4,SetVar(CFU_MESSAGE=)
exten = 2120,5,SetVar(CFU_FLAGS=)
exten = 2120,6,SetVar(CFU_LIMIT=)
exten = 2120,7,SetVar(DIAL_DEST=Zap/2)
exten = 2120,8,SetVar(DIAL_TIMEOUT=15)
exten = 2120,9,SetVar(DIAL_FLAGS=)
exten = 2120,10,SetVar(DIAL_LIMIT=1)
exten = 2120,11,SetVar(CFNA_DEST=)
exten = 2120,12,SetVar(CFNA_TIMEOUT=)
exten = 2120,13,SetVar(CFNA_MESSAGE=)
exten = 2120,14,SetVar(CFNA_FLAGS=)
exten = 2120,15,SetVar(CFNA_LIMIT=1)
exten = 2120,16,SetVar(CFBL_DEST=)
exten = 2120,17,SetVar(CFBL_TIMEOUT=)
exten = 2120,18,SetVar(CFBL_MESSAGE=)
exten = 2120,19,SetVar(CFBL_FLAGS=)
exten = 2120,20,SetVar(CFBL_LIMIT=)
exten = 2120,21,SetVar(OPER_DEST=${COLISEUM}/2221${ORMOND}/95042325606)
exten = 2120,22,SetVar(OPER_TIMEOUT=20)
exten = 2120,23,SetVar(OPER_MESSAGE=)
exten = 2120,24,SetVar(OPER_FLAGS=)
exten = 2120,25,SetVar(OPER_LIMIT=)
exten = 2120,26,Macro(std-exten)
[macro-std-exten]
;
; This macro is controlled by the following variables set before the macro
; is called. The variables can be set using SetVar, DBGet, AGI, web based
; CGI script, manager interface, government mind control rays, etc. How
; they are set is up to you.
;
; MOH_CLASS - Music on Hold Class. Set the music on hold class to this
; value. If empty, don't change the current MOH class.
;
; DND - Do Not Disturb. If this variable is not empty, then send call
; directly to unavailable voicemail. Example: SetVar(DND=1) or
; SetVar(DND=blah) or SetVar(DND=)
;
; DND_MESSAGE - When DND is set, play this audio file before going to
; unavailable voicemail. Example:
SetVar(DND_MESSAGE=/tmp/happy-message) or
; SetVar(DND_MESSAGE=)
;
; CFU_DEST - Call Forward Unconditional. If set, unconditionally dial
; this destination. If empty, don't Call Forward Unconditionaly.
Example:
; SetVar(CFU_DEST=IAX2/[EMAIL PROTECTED]/1234) or SetVar(CFU_DEST=)
;
; CFU_TIMEOUT - When dialing CFU_DEST, how many seconds to wait before
; giving up and going to unavailable voicemail. If empty or 0, never
; timeout. Example: SetVar(CFU_TIMEOUT=20)
;
; CFU_MESSAGE - When dialing CFU_DEST, play this audio file to the caller
; first. If empty, don't play anything. Example:
; SetVar(CFU_MESSAGE=/tmp/happy-message) or SetVar(CFU_MESSAGE=)
;
; CFU_FLAGS - When dialing CFU_DEST, use this variable as the optons on to
; the Dial command. If empty, use the default Dial flags. Example:
; SetVar(CFU_FLAGS=mH) or SetVar(CFU_FLAGS=)
;
; CFU_LIMIT - When dialing CFU_DEST, this is the maximum number of
; simultaneous calls to allow. If empty or 0, allow unlimited
simultaneous
; calls. Example: SetVar(CFU_LIMIT=1) or SetVar(CFU_LIMIT=)
;
; DIAL_DEST - If DND and CFU_DEST are not set, dial this destination. If
; empty go to unavailable voicemail. If all the destinations are busy
dial
; CFBL_DEST. Example: SetVar(CFU_DEST=IAX2/[EMAIL PROTECTED]/1234) or
; SetVar(CFU_DEST=SIP/1234Zap/4)
;
; DIAL_TIMEOUT - How many seconds to wait before giving up and trying
; Call Forward No Answer. If empty or 0, never timeout. Example:
; SetVar(DIAL_TIMEOUT=20)
;
; DIAL_FLAGS - Use this variable as the optons on to the Dial command. If
; empty, use the default Dial flags. Example: SetVar(DIAL_FLAGS=mH) or
; SetVar(DIAL_FLAGS=)
;
; CFNA_DEST - Call Forward No Answer. If DIAL_TIMEOUT expires when
; dialing DIAL_DEST (call was not answered) then dial this destination. If
; empty, don't Call Forward No Answer and go directly to unavailable
; voicemail. If all destinations are busy go directly to unavailable
; voicemail. Example: SetVar(CFNA_DEST=IAX2/[EMAIL PROTECTED]/1234) or
; SetVar(CFU_DEST=SIP/1234Zap/4) or SetVar(CFNA_DEST=)
;
; CFNA_TIMEOUT - When dialing CFNA_DEST, how many seconds to wait before
; giving up and going to unavailable voicemail. If empty or 0, never
; timeout. Example: SetVar(CFNA_TIMEOUT=20)
;
; CFNA_MESSAGE - Before dialing CFNA_DEST, play this audio file to the
; caller first. If empty, don't play anything. Example:
; SetVar(CFNA_MESSAGE=/tmp/happy-message) or SetVar(CFNA_MESSAGE=)
;
; CFNA_FLAGS - When dialing CFNA_DEST, use this variable as the optons on
; to the Dial command. If empty, use the default Dial flags. Example:
; SetVar(CFNA_FLAGS=mH) or SetVar(CFNA_FLAGS=)
;
; CFNA_LIMIT - When dialing CFNA_DEST, this is the maximum number of
; simultaneous calls to allow. If empty or 0, allow unlimited
simultaneous
; calls. Example: SetVar(CFNA_LIMIT=1) or SetVar(CFNA_LIMIT=)
;
; CFBL_DEST - Call Forward Busy Line. If destination is busy when dialing
; DIAL_DEST then dial this destination. If empty, don't Call Forward Busy
; Line and go directly to busy voicemail. Example:
; SetVar(CFBL_DEST=IAX2/[EMAIL PROTECTED]/1234) or
SetVar(CFBL_DEST=SIP/1234Zap/4)
; or SetVar(CFBL_DEST=)
;
; CFBL_TIMEOUT - When dialing CFBL_DEST, how many seconds to wait before
; giving up and going to busy voicemail. If empty or 0, never timeout.
;
[Asterisk-Users] T100P Frame Errors
I have been messing with the T100P card with and without data for over a week now, and still to no avail. Just got off the phone with our T1 provider to make sure our settings were correct for the T1 in zaptel. zaptel.conf: span=0,1,0,esf,b8zs nethdlc=1-20 fxsks=21-28 loadzone = us defaultzone=us ifconfig just after one traceroute attempt to our provider dns server ip hdlc0 Link encap:(Cisco)-HDLC inet addr:160.81.118.46 P-t-P:160.81.118.45 Mask:255.255.255.252 UP POINTOPOINT RUNNING MTU:1500 Metric:1 RX packets:0 errors:415 dropped:0 overruns:0 frame:415 TX packets:474 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:10 RX bytes:0 (0.0 b) TX bytes:8766 (8.5 Kb) loLink encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:42 errors:0 dropped:0 overruns:0 frame:0 TX packets:42 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:3068 (2.9 Kb) TX bytes:3068 (2.9 Kb) My T1 provider seems to think the transmit signal of the T100p is not strong enough and is the cause of the frame errors on the RX trip. The cable works fine with our cisco so I don't think it is a faulty cable. The cable is a standard ethernet cable (which is supposed to work according to digium tech support). I get the same response when i use a t1 xover 1-4 , 2-5 (rj48c) I have even tried an rj48s but that did not five me any lights on the card. Connecting T100P to HyperEdge DTWA-528-02 Smart Jack (provided by verizon) Has anybody any input or possible resolution? Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] CHANUNAVAIL = CHANUNAVAIL doesn't eval properly
exten = _3XXX,102,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?i,1:103)
Matthew Boehm wrote:
Here is the relevant dialplan:
exten = _3XXX,1,Dial(SIP/${EXTEN},15,tr)
exten = _3XXX,2,Voicemail([EMAIL PROTECTED])
exten = _3XXX,102,GotoIf($[${DIALSTATUS}==CHANUNAVAIL]?i,1:103)
exten = _3XXX,103,Voicemail([EMAIL PROTECTED])
exten = i,1,Playback(invalid)
exten = i,2,Hangup()
What 'should' happen is that if someone dials any extension starting with 3,
Dial attempts to dial it. If there is no such channel, set DIALSTATUS and
goto priority n + 101. Then check result of DIALSTATUS. If DIALSTATUS is
equal to CHANUNAVAIL then goto extension 'i' else goto priority 103.
Here is an attempt to dial non-existant extension 3652:
Oct 15 13:10:06 WARNING[1116730816]: chan_sip.c:1384 create_addr: No such
host: 3652
Oct 15 13:10:06 NOTICE[1116730816]: app_dial.c:742 dial_exec: Unable to
create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing GotoIf(SIP/3044-9a9c,
CHANUNAVAIL=CHANUNAVAIL?cytel-internal|i|1:103) in new stack
-- Goto (cytel-internal,3652,103)
-- Executing VoiceMail(SIP/3044-9a9c, [EMAIL PROTECTED]) in new stack
Oct 15 13:10:06 WARNING[1116730816]: app_voicemail.c:1367 leave_voicemail:
No entry in voicemail config file for '3652'
It seems that CHANUNAVAIL does not equal CHANUNAVAIL in this case. This is
incorrect. Any ideas?
Thanks,
Matthew
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begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
x-mozilla-html:FALSE
version:2.1
end:vcard
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[Asterisk-Users] New asterisk user question
Hi, I'm going to setup asterisk as a voip gateway for remote internet users. I'm going to use cisco 2600 for it with E1 interface cards. And I have a few questions. 1. My provider will provide me with a couple of real phone numbers (MSN it's called iirc), is there a way to assign these numbers to voip clients (preferably ip phones)? I mean, will it be possible to dial in to these users? 2. Has anybody been running asterisk on FreeBSD/Linux on a smp amd64 server? Does an asterisk server do a heavy load? 3. Can anybody propose ip phones cheaper than cisco's or avaya's ones? -- Micha Nasiadka [EMAIL PROTECTED] ___ 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] T100P Frame Errors
On Fri, 2004-10-15 at 14:25 -0400, Cirelle Enterprises wrote: I have been messing with the T100P card with and without data for over a week now, and still to no avail. Just got off the phone with our T1 provider to make sure our settings were correct for the T1 in zaptel. zaptel.conf: span=0,1,0,esf,b8zs Spans are 1 based, not zero. span=1,1,0,esf,b8zs nethdlc=1-20 fxsks=21-28 loadzone = us defaultzone=us -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Sample advanced call routing standard extension
On Fri, 15 Oct 2004 13:26:12 -0500, Eric Wieling [EMAIL PROTECTED] wrote: [default] ; ; Eric Wieling Eric, Great stuff! I wish more people would post their configs. A lot can be learned from examples. Maybe find a home on the wiki for this! -Chuji ___ 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
[Asterisk-Users] Should ZAP channels pass CNAM to SIP?
signalling=pri_cpe callerid=asreceived I see that I get the callerID CNAM in the cdr records, but the same information does not show up on the display on my Cisco 7960 phone only the ANI. I do get Callerid from voip to voip calls . Just not on the zap to voip calls. My question is does anyone have CNAM passing through to voip? I am trying to figure out if I have a configuration error or this is a system limitation. Sure would be nice to see the name of the weirdos that call me. ___ 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] Prerelease of DIAX 0.9.9a
Brian West wrote: Anyway we could talk you into releasing the source? I would love to see wider codec support. And the ability to launch the URL sent with the IAX call. Brian, The codec stuff I did, and the source is all available at iaxclient.sf.net. Afaik, all the existing IAX softphones use iaxclient except for firefly [and maybe this new one I heard about]. Iaxclient is LGPL. I just made the codec system modular, so writing your own codec driver now takes about an hour. Already included are GSM, uLaw, speex, and ilbc. I was going to add lpc-10 yesterday, but it seems that it's a bit of a PITA to do without copying from asterisk's lpc10 driver implementation (which is GPL). I'm actually planning on adding the speex settings stuff in an API as well.. -SteveK ___ 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
[Asterisk-Users] calling out from a remote * server
I have set up 2 * servers and connected them via IAX2, the connection
works, so far so good.
To optimize on the phone bill however I would like to have calls
that are local for the remote * server placed through the remote server.
How is this accomplished? I first tried the manual approach (dialing an
8 would make the call go through the remote * server) but it doesn't work,
the call is still placed from the local server.
This is what I put in my extensions.conf:
[remote-out]
switch = IAX2/user:[EMAIL PROTECTED]/pstn-local
exten = _8.,1,Dial(ZAP/g1/${EXTEN:1},70,T)
exten = _1NXXNXX,1,Dial(ZAP/g1/${EXTEN})
exten = _NXX,1,Dial(ZAP/g1/${EXTEN})
exten = _8.,2,Macro(fastbusy)
Ideally I would also like * to strip the area code if the remote server is
used (it's a local call then) but this is detail. Ultimately I would like
to do the same with international calls.
I couldn't find the solution in the wikis.
Thanks all!
___
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RE: [Asterisk-Users] Re: Advice on OS Choice
On Fri, 2004-10-15 at 16:11 +0100, Kevin Walsh wrote: Jason T. Nelson [EMAIL PROTECTED] wrote: In our last exciting episode, Kevin Walsh ([EMAIL PROTECTED]) said: If GNU/Linux was licensed under a BSD-style license then Red Hat could easily close the source - just as Apple did when they stole BSD code to create their OS/X effort. I don't believe that Red Hat would do that sort of thing anyway - those tactics are best left to Apple and Microsoft. Umm.. subtle but very important point here. Apple did not steal BSD code. BSD code cannot be stolen. It is given away as basically a gift. Stealing implies that the person you stole from has now lost something. Perhaps steal was a bit harsh then. Maybe I should have said Apple, Microsoft and others close the source with no compensation nor recognition given to the original authors, as allowed by the stupid BSD license. It's the authors' fault really. They live and learn. Perhaps they'll use the GPL next time. Please do not call the BSD license stupid. It has a reason. While I don't like the connotation of the statement that the GPL is viral, it is true. BSD licensing allows you to take portions of code private for interspersing with proprietary code. Both have their basis in beliefs about the nature of people. Person opinions only here but the GPL seems to be pessimistic about the nature of people and companies. It forces the software it intermingles with to also be open source. And that by legally forcing code distributions to be open sourced, it gives the community a greater chance of getting the derivative works either directly or via a person willing to purchase the product and share. BSD seems to be very optimistic in that there will be enough people willing to contribute back to a project that the amount not contributed back won't matter much. In the business I work for, we have to be careful of the licenses we use like everyone else. We keep our eye towards making a profit off of some of our code, and we are always looking for ways to make money off of services. Some of our code is BSD licensed and held privately. Some of our code is held in proprietary licenses. We use GPL software when it doesn't cause trouble with our motivations. We tend to contribute to all we can. If we BSD license our code we wish to share as a company, we don't ever have to worry about someone elses changes causing us trouble in our proprietary code. We then get the best of both open source collaberation and proprietary code. All that and I contribute to both GPL and BSD licensed projects. Neither side is stupid, just differently motivated. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
