Re: [Asterisk-Users] aastra 480i config files
On Sun, 2005-11-20 at 20:52 -0600, Michael Graves wrote: Would anyone on-list be able to provide me with some sample config files for the Aastra 480i SIP phone. I'd like to to migrate from individually hand tweaked to centrally FTP provisioned, but need somewhere to start. I'm also looking to see how others handle multiple call appearances. Well I got all the info from Aastra's examples in their admin guide for my Aastras. Have you looked at http://www.aastra.com/enterpriseip/pro_239.asp?view=downloads -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Eicon Diva Server query
Yes, you can use the Eicon Diva Range with 2.6 Kernels See this page to see how to get an Eicon Diva working with Asterisk. http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlasis Hatzistavrou - asterisk mailing list account Sent: 18 November 2005 16:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Eicon Diva Server query Avi Miller wrote: Hello gurus! I've given up on crappy passive ISDN cards and am heading into the wild world of real, Active Super Dooper Server boards. I have a choice of two Eicon Diva Server cards: Eicon Diva Server 4BRI Eicon Diva Server V-4BRI Hello, We've been using an Eicon Diva Server 4BRI with a RH 9 installation (kernel 2.4.20-8). It works great in both TE and NT mode. I assume that it will work equally great with a 2.6 kernel... Best regard, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Death at 2am
Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. In /var/log/messages I just see endless entries like this: Nov 21 02:00:13 WARNING[18841] chan_sip.c: No such host: voipfone.co.uk Nov 21 02:00:13 WARNING[18841] chan_sip.c: Probably a DNS error for registration to [EMAIL PROTECTED], trying REGISTER again (after 20 seconds) An attempt to connect to the console leaves asterisk eating up CPU cycles and this in /var/log/messages Nov 20 11:51:25 WARNING[94218] asterisk.c: Accept returned -1: Too many open files A message which reoccurs several hundred times a second. Can anyone either solve this problem for me completely, or at least give me a hint as to the significance of 02:00? Is this an Asterisk thing (most of my configurations are as per install samples), or an underlying OS thing? Thanks -- Chris Hastie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register redirect
Matt Riddell wrote: Marc Storck wrote: Hello, I would like to know if there is a way in IAX2 and SIP to tell a client to register at a different server. For example: Client tries to register at server B but server B answers with some sort of redirect to tell the client to register at server C. The client then tries to register with Server C. In theory we could send a SIP 302 redirect for REGISTER, but when I checked this one year ago, no phones understood this. It is covered in the RFC though. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problems with Read() in outgoing calls
John Biundo [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I posted the following a couple of days ago. My problem was inbound, but the workaround might be worth a try: == Bug or feature? Thanks John, It appears that my outbound supplier has a different default DTMF setting for outbound and inbound, which is where the problem was. Kind Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme + sendtext
BJ Weschke wrote: On 11/19/05, Jean-Denis Girard [EMAIL PROTECTED] wrote: Hi all, Is sending text to a conference supported by asterisk-1.2, ie one member of the conference sends text, it is received by all other members of the conference (provided their channel supports text of course) ? I made a quick test with IAX softphones, and it seems that text isn't sent to IAX channels through the conference. If it is not supported, how difficult would that be to add this functionnality ? It's not supported at this time, but you're also not the first person to ask about it. I suppose it wouldn't be too difficult to add. Well, we have no routing of incoming text messages through the dial plan, but I guess this is a bit more simple, since we have open channels into meetme. /o ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: /spool/outgoing delays
Matt Riddell [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Chris Cahill wrote: The process then goes on to call a few agi scripts, and ends up creating another file (via agi) in the outgoing directory, this one being the one that calls the outside world. Are you *creating* the file in the /outgoing directory? You should create it somewhere else and move it into /outgoing, to prevent asterisk to find an incomplete file. Leif Leif, Thanks for your suggestion, but yes I am creating it elsewhere and moving it in. Does it always have a unique filename? Indeed it does!! C ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mISDN and chan_isdn for 1.2
John Martin wrote: Can anyone recommend a version of mISDN and mISDNuser (dates of CVS or archive held on someone’s server) that will work with the chan_isdn in Asterisk 1.2. I have used the install-misdn script on http://www.beronet.com/download/ and that seems to work.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel compilation help!
Hi, I'm compiling Zaptel1-1.0.9 in Sparc64/Debian and I'm getting these errors. I compiled asterisk on the same machine and it went ok. I want to activate the conference feature of asterisk thats why i'm compiling zaptel. These are the errors: sip:/usr/local/src/zaptel-1.0.9.1# make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/include/linux/dcache.h:10, from /usr/include/linux/fs.h:17, from /usr/include/linux/proc_fs.h:6, from zaptel.c:45: /usr/include/linux/rcupdate.h: In function `rcu_pending': /usr/include/linux/rcupdate.h:114: error: invalid lvalue in unary `' /usr/include/linux/rcupdate.h:116: error: invalid lvalue in unary `' /usr/include/linux/rcupdate.h:117: error: invalid lvalue in unary `' zaptel.c: In function `zt_register': zaptel.c:4406: warning: implicit declaration of function `class_simple_device_add' zaptel.c: In function `zt_unregister': zaptel.c:4456: warning: implicit declaration of function `class_simple_device_remove' zaptel.c: In function `zt_init': zaptel.c:6431: warning: implicit declaration of function `class_simple_create' zaptel.c:6431: warning: assignment makes pointer from integer without a cast zaptel.c: In function `zt_cleanup': zaptel.c:6492: warning: implicit declaration of function `class_simple_destroy' make: *** [zaptel.o] Error 1 please help, Thanks, Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Death at 2am
In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. In /var/log/messages I just see endless entries like this: Nov 21 02:00:13 WARNING[18841] chan_sip.c: No such host: voipfone.co.uk Nov 21 02:00:13 WARNING[18841] chan_sip.c: Probably a DNS error for registration to [EMAIL PROTECTED], trying REGISTER again (after 20 seconds) An attempt to connect to the console leaves asterisk eating up CPU cycles and this in /var/log/messages Nov 20 11:51:25 WARNING[94218] asterisk.c: Accept returned -1: Too many open files A message which reoccurs several hundred times a second. Can anyone either solve this problem for me completely, or at least give me a hint as to the significance of 02:00? Is this an Asterisk thing (most of my configurations are as per install samples), or an underlying OS thing? Firstly, look and see what the too many open files are: # lsof -p94218 (or whatever the complaining PID is). I'm assuming FreeBSD has lsof. I don't know, as I use Linux. Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. You could also try specifying 212.187.162.178 temporarily instead of voipfone.co.uk - that would tell you whether the problem is DNS related. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 Gateway
Hi all! Someone who can recommend a good E1 gateway for terminating VoIP traffic. H323 or Sip Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Well that didn't work. When I rebooted MySQL didn't start at allOn 11/21/05, JP Carballo [EMAIL PROTECTED] wrote:JP Carballo wrote: Eric Bishop wrote: I have: [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql mysqld0:off 1:off 2:off 3:on4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk asterisk0:off 1:off 2:on3:on4:on5:on6:off What would you suggest I do? snip rant Holy crap, this kind of replying is getting me dizzy! Up, down, what next? Left and right? Why can't we just agree to delete all previous text, anyway we all have threaded readers...don't we? /rant chkconfig --level 3 mysqld off chkconfig --level 2 mysqld on chkconfig --level 2 asterisk offI forgot to add that you should get this:([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep asterisk\|mysqldmysqld 0:off1:off2:on3:off4:off5:off6:offasterisk 0:off1:off2:off3:on4:off5:off6:off--JP Carballohttp://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to Fax Server
Title: Message Hi I'm looking for following solution: Asterisk is connected to PSTN by Digium or some another card which has Fax Detection If incoming call is a fax I woud like to transfer it to External Fax server by SIP or H323 for getting a Fax. If incoming call is a voice to direct it to another trunk. Is it possible to make it on Asterisk? If yes which E1 card is preferable? Thanks in advance Arcady ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 Gateway
Anders Svensson wrote: Someone who can recommend a good E1 gateway for terminating VoIP traffic. H323 or Sip Asterisk! /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to Fax Server
Title: Message Hi I'm looking for following solution: Asterisk is connected to PSTN by Digium or some another card which has Fax Detection If incoming call is a fax I woud like to transfer it to External Fax server by SIP or H323 for getting a Fax. If incoming call is a voice to direct it to another trunk. Is it possible to make it on Asterisk? If yes which E1 card is preferable? Thanks in advance Arcady ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Well that didn't work. When I rebooted MySQL didn't start at all The level doesn't set _when_ something starts, just _if_ something starts. Some daemons should start in single user mode, some not. Some others should only start when in GUI mode, others not, etc. This is what level controls. When something starts is usually controlled with the name of the start/stop scripts in /etc/rcx.d/ (or something like that). Files starting with the S00 prefix are started first, files with S99 are started last for that runlevel. The same for K00 and K99, but that describes the time when processes are killed. So if Asterisk is started using S80asterisk, and MySQL using S50mysqld, then it obviously isn't going to work as intended. The same also when both are started with S99, because asterisk will be started before mysqld... I usually mess around with the numbers, but that is not very reproducable, dependencies listed in the rpm file (or equivalent) usually takes care of this. When isntalling from source, you're on your own. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bluetooth headset with softphone or direct asterisk
On Sat, 2005-11-19 at 13:47 -0800, Ben Higley wrote: [AG] Pocket_PC AT+BRSF=23 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CIND=? [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CIND? [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CMER=3,0,0,1 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CLIP=1 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CGMI=? [AG] Pocket_PC ERROR Strange behaviour. Do you get similar behaviour if you connect to the phone with minicom? Are you connecting on the correct channel? Show sdptool browse output for the phone. -- dwmw2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with multiplier
Hi all, i've a problem on a PSTN line that i've connected to Asterisk server with a Diginum Card. A diagram: +--[Fax Machine] | [PSTN]+ | +--[Asterisk] The problem is if i pick up the phone on Fax Machine i get no Dialtone. If i disconnect asterisk cable all goes ok. I think the problem is because Diginum Card detect the off hook and take the line. Can i solve this ? Thanks, Michele -- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk startup
On 21 Nov 2005, at 00:38, Luki wrote:LD_ASSUME_KERNEL 2.4.1 ... will make the kernel do old-styleprocess-perthread posix threads. I don't have this anywhere in the startup script on 2.6.12-1.1372_FC3and still have only one process in ps:Sorry, I wasn't clear, if you _do_ have LD_ASSUME_KERNEL 2.4.1in your start-up script(or a real 2.4.1 kernel) you will get multiple lines in your ps output,one per thread.If you have a newer kernel and _don't_ have LD_ASSUME_KERNEL 2.4.1set you will get one line in your ps output - all the threads running ina single process.$ ps aux|grep asteriskasterisk 4649 0.0 0.2 17744 1080 ? Sl Sep08 37:19/usr/sbin/asterisk -U asterisk -G asterisk -g -v -nIt's Sunday, it's quiet, no calls currently. Not that it should make adifference, but I run asterisk in a chrooted environment with its owncopy all all shared libs, etc. Maybe it is running in the old-stylemode, but I don't know how to check for sure. Oh well, still amystery...No, what you see agrees with what I was trying to say , I just didn'texpress my self that well :-) (hey it was sunday night)Tim. http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth and Ericcson T68 problem
Hello Enky, We have encountered similar problems with various Ericsson Nokia phones. We couldn't get the channel driver to work 100%. However, we cannot actually tell whether it was our mistake or whether there was a problem with the channel driver. We tried to contact the driver's maintainer/creator but no luck... If you manage to find a solution for this problem we'd also be interested to know about it. Best regards, Vlasis. Enky wrote: Hi, I have read many pages and tried many things, but without any success. I have paired my ERICCSON T68 with the Asterisk PC. The Asterisk version is “Asterisk CVS-v1-0-11/19/05-14:52:52”. The chan_bluetooth is the last release, downloaded from “http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz”. It is all OK. I can dial from the Asterisk a number. The T68 dials it, but when the called party picks the phone and the call goes connected there is no any audio! Neither from or to the Asterisk. Here are a short logs: This is the initial log, when I start the Asterisk and it connects the T68. It seems OK: ---cut--- Asterisk Ready. *CLI Nov 19 15:15:45 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? Nov 19 15:15:46 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:417 set_cind: Audio Gateway T68 got signal [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- This is when I dial a number. It seems OK too, but no audio when connects: ---cut--- -- Executing Dial(SIP/222-3885, BLT/T68/123|60) in new stack [AG]T68 ATD123; -- Called T68 [AG]T68 OK [AG]T68 +CIEV: 8,1 -- BLT/T68 answered SIP/222-3885 [AG]T68 +CIEV: 2,4 [AG]T68 +CIEV: 2,5 ---cut--- And this is when I interrupt the dialed call: ---cut--- [AG]T68 AT+CHUP == Spawn extension (default, 2002, 1) exited non-zero on 'SIP/222-3885' [AG]T68 OK Nov 19 15:18:06 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2493 rd_close: Device T68 disconnected, scheduled reconnect in 5 seconds: Connection reset by peer (errno 104) Nov 19 15:18:11 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- Please someone to help me :) Thank you in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] calling to asterisk and listening to music (GSM) --Anyone, please?????
Hi all! I'm trying to play some music from asterisk, and when I call to the PBX from a GSM mobile phone, the more I speak while hearing the music, the worst is the quality of the music I hear... My audio is at 8Khz, 16bits/sample. I've tried different codecs for asterisk, but results are the same... If I call to the PBX from a conventional phone, I can speak while hearing the music, with no quality loss... Hi, try the following: Take one phone and hodl it next to a normal speaker with music. Now call it with a mobile phone and try the same. It results for me in the same: The music is disturbed. So I am sure it has to do with the mobile GSM-standard and not with asterisk. Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 Gateway
I dont think its a good idea to put an * in Bosnia when we are in Sweden. Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: den 21 november 2005 10:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] E1 Gateway Anders Svensson wrote: Someone who can recommend a good E1 gateway for terminating VoIP traffic. H323 or Sip Asterisk! /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure the LCS with Asterisk---Anyone, please?????
Hi, I am trying to configure the LCS with the Asterisk1.0.3 Is that needed to modify some code in configuration files or it needed to modify in the source code too. Can anyone please suggest how to configure the LCS? Regards, Bidyut Wipro Technologies Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or [EMAIL PROTECTED] immediately and destroy all copies of this message and any attachments. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 Gateway
[EMAIL PROTECTED] wrote: I dont think its a good idea to put an * in Bosnia when we are in Sweden. Why not? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you get a sound to play to caller on answer?
I tried this dial command to get a sound to play to the caller on answer. I have even tried to use the LIMIT_CONNECT_FILE option with no success. As can be seen below the start_sound variable shows 'UNDEF'. Are there some other settings I have missed out, eg. file location, type etc. The sound file is in GSM format. SIP/providername/002345678|42|HL(2658:61000:3:LIMIT_CONNECT_FILE=soundfile) -- Limit Data: -- timelimit=2658 -- play_warning=61000 -- play_to_caller=yes -- play_to_callee=no -- warning_freq=3 -- start_sound=UNDEF -- warning_sound=timeleft -- end_sound=UNDEF Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Death at 2am
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. Firstly, look and see what the too many open files are: # lsof -p94218 (or whatever the complaining PID is). I'm assuming FreeBSD has lsof. I don't know, as I use Linux. Thanks. I'll give this ago tomorrow morning Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. I did look at that. Nothing seems to run at 2am that doesn't run on every other hour. My first inclination was that maybe newsyslog was trying to rotate Asterisk's logs at 2am, but that doesn't look to be the case. I take it that Asterisk (with mostly the standard sample config files) doesn't try to do anything at 02:00 then? Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. I hadn't considered that. Connectivity provider, or VOIP provider (of the latter I have more than one)? I'll experiment with some debug output overnight tonight to see if it gives me any more clues. You could also try specifying 212.187.162.178 temporarily instead of voipfone.co.uk - that would tell you whether the problem is DNS related. I'm pretty sure it is not DNS related. Asterisk seems to loose the ability to connect to anything, irrespective of the direction of the connection. Phones can not connect to Asterisk either. I think the inability of Asterisk to connect to a DNS server is merely one of the symptons of a total inability to talk to anything else at all. -- Chris Hastie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Death at 2am
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. Firstly, look and see what the too many open files are: # lsof -p94218 (or whatever the complaining PID is). I'm assuming FreeBSD has lsof. I don't know, as I use Linux. Thanks. I'll give this ago tomorrow morning Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. I did look at that. Nothing seems to run at 2am that doesn't run on every other hour. My first inclination was that maybe newsyslog was trying to rotate Asterisk's logs at 2am, but that doesn't look to be the case. I take it that Asterisk (with mostly the standard sample config files) doesn't try to do anything at 02:00 then? Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. I hadn't considered that. Connectivity provider, or VOIP provider (of the latter I have more than one)? I'll experiment with some debug output overnight tonight to see if it gives me any more clues. You could also try specifying 212.187.162.178 temporarily instead of voipfone.co.uk - that would tell you whether the problem is DNS related. I'm pretty sure it is not DNS related. Asterisk seems to loose the ability to connect to anything, irrespective of the direction of the connection. Phones can not connect to Asterisk either. I think the inability of Asterisk to connect to a DNS server is merely one of the symptons of a total inability to talk to anything else at all. -- Chris Hastie ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 Gateway
-Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Monday, November 21, 2005 5:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] E1 Gateway [EMAIL PROTECTED] wrote: I dont think its a good idea to put an * in Bosnia when we are in Sweden. Why not? I put one in Senegal when I am in Washington DC. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is there any free pocket pc softphone??
I was able to register Portrait with our Asterisk box, but no audio, no signaling at all. Played a while with different codecs but no success. Did anybody make it really work with asterisk? Any hints, configs etc. Regards Guido Hecken I've also had some luck with Microsoft Portrait Guido Hecken wrote: is there any free pocket pc softphone Try sjphone from http://www.sjlabs.com/sjp.html Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] customized softphones
Im looking for SIP phone *** REPLY SEPARATOR *** On 11/20/2005 at 8:01 AM Time Bandit wrote: Hi there, is there any free softphone that i can customize accoring to my needs ?? You could use IaxComm : http://iaxclient.sourceforge.net/iaxcomm/index.html hth ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] ztdummy problem on SUSE 9.3
On Saturday 19 November 2005 19:12, dolcicbe wrote: Hello can you explain me that more exactly. Thank you Bernhard I think what is meant is that SuSe already come with zaptel modules in /lib/modules/`uname -r`/extra The zaptel build process puts the its modules in /lib/modules/`uname -r`/misc and when you load the modules 'modprobe' will pick up the wrong version. See http://www.voip-info.org/wiki-Asterisk+Linux+SuSE scroll down to the section on suSE 9.2 Paul _ Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mustafa N. Deeb Gesendet: Samstag, 19. November 2005 12:46 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Betreff: RE: [Asterisk-Users] ztdummy problem on SUSE 9.3 Hi That's b/c the make install command inserted in a directory different than the one configured in modprobe.conf.. Cheers _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dolcicbe Sent: Saturday, November 19, 2005 1:25 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy problem on SUSE 9.3 Hello I'm a beginner and I want to install the meetme module for asterisk and therefor it is necessary to install ztdummy. I have a lot of problems, always comes the error stated down. I hope somebody already have got the solution and is willing to help me. Greeting from Austria Bernhard gl0:/usr/src/zaptel-1.0.9.2 # modprobe zaptel gl0:/usr/src/zaptel-1.0.9.2 # modprobe ztdummy FATAL: Error inserting ztdummy (/lib/modules/2.6.11.4-20a -default/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) gl0:/usr/src/zaptel-1.0.9.2 # dmesg |tail load_module: err 0xfffe (dont worry) ztdummy: disagrees about version of symbol zt_receive ztdummy: Unknown symbol zt_receive, st_info == 0x1 ztdummy: disagrees about version of symbol zt_transmit ztdummy: Unknown symbol zt_transmit, st_info == 0x1 ztdummy: disagrees about version of symbol zt_unregister ztdummy: Unknown symbol zt_unregister, st_info == 0x1 ztdummy: disagrees about version of symbol zt_register ztdummy: Unknown symbol zt_register, st_info == 0x1 load_module: err 0xfffe (dont worry) -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can not build zaptel with kernel-2.6.12
Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can not build zaptel with kernel-2.6.12
harry gaillac wrote: Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // I have to ask the obvious question. Do you have the same source as you have kernel running? Remember if you have run an upgrade it could have updated the kernel but may not have doen the sources and if you have the sources from the installion media then you would have different versions that will cause this exact problem. David Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Once upon a time Sunday 20 November 2005 10:38 pm, JP Carballo wrote: JP Carballo wrote: Eric Bishop wrote: I have: [EMAIL PROTECTED] ~]# chkconfig --list | grep mysql mysqld 0:off 1:off 2:off 3:on4:off 5:off 6:off [EMAIL PROTECTED] ~]# chkconfig --list | grep asterisk asterisk0:off 1:off 2:on3:on4:on5:on6:off What would you suggest I do? snip rant Holy crap, this kind of replying is getting me dizzy! Up, down, what next? Left and right? Why can't we just agree to delete all previous text, anyway we all have threaded readers...don't we? /rant chkconfig --level 3 mysqld off chkconfig --level 2 mysqld on chkconfig --level 2 asterisk off I forgot to add that you should get this: ([EMAIL PROTECTED]:asterisk)# chkconfig --list | grep asterisk\|mysqld mysqld 0:off1:off2:on3:off4:off5:off 6:off asterisk 0:off1:off2:off3:on4:off5:off 6:off ok a little back round on runlevels. Linux allows for up to 10 runlevels, 0-9, but usually only some of these are defined by default. Runlevel 0 is defined as ``system halt''. Runlevel 1 is defined as ``single user mode''. Runlevel 6 is defined as ``system reboot''. Other runlevels are dependent on how your particular distribution has defined them, and they vary significantly between distributions. Looking at the contents of /etc/inittab usually will give some hint what the predefined runlevels are and what they have been defined as. ok so when you turn mysqld off on run level 3 and thats what you system runs as mysqld will never start. the services selected for that run level are ran when you enter that run level. the order that they are run at is defined by a priority system. so you need to make sure the priority of asterisk is such that is it started after mysqld. on my system mysqld has a priority of 64 and asterisk is 99 look in /etc/rc3.d the files starting with a S are for startup and K for shutdown. they start with lowest number up through highest number. that last thing ran is /etc/rc.local so you could always put in there /etc/init.d/asterisk restart to make sure its the last thing done. -- Dennis Gilmore, RHCE dennis AT ausil DOT us http://www.ausil.us pgp3NECiMWqae.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
Once upon a time Sunday 20 November 2005 8:39 pm, Matt Riddell wrote: Eric Bishop wrote: Hi All, I am running Asterisk (1.0.9.) on CentOS 4 with CDR recording being output to MySQL. However whenever the system boots up after a reboot I am needing to manually restart Asterisk because MySQL is after Asterisk in the service startup sequence and I get ERROR[3367]: Failed to connect to mysql database cdr on localhost. Anyone know of a simple and elegant way to fix this? I'd prefer not to have to hack either MySQL or Asterisk init scripts If it's running using services, you could set MySQL to start on level 2 and Asterisk on level 3. chkconfig --list umm. you obviously dont understand how the different run levels work. run level 2 has nothing to do with run level 3 the easiest way would be to put in /etc/rc.local /etc/init.d/asterisk restart then asterisk will be restarted very last thing before you get a login prompt. that is the only way to do it without changing the priorites in the init scripts to make sure asterisk starts later. though on my setup my init scripts are set to run asterisk almost last and way after mysql has started. -- Dennis Gilmore, RHCE dennis AT ausil DOT us http://www.ausil.us pgpFqyDCOtbRZ.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User identification
Hi! I'm new to asterisk and I'm trying to develope an application that allows the caller to input an Id and then the system redirects to an operator. What I did so far is to create an agi script that receives the call, allow the input, query a DB to check thah the Id is valid and then transfer. I need advice on the way that the application that uses the operator can process the input Id. Change the callerid? other options? -- Best Regards, Ezequiel Gonzalez Rial ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP channels
Hi all, i've a problem in my Asterisk system. We have around 30 SIP phones connected to an asterisk system, and sometimes some SIP channel (associated to an extension) gets busy all the time, even whenthat extensionisn't in use. We have a workaround for this, as we can't restart asterisk in work hours, we assign other extension to that phone and create an alias for calling there. When asterisk is restarted, all extensions aswer the way it's supposed to be. Any clues? Thank you :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12
Salut Harry, plus de nouvelles de toi :( Serais tu faché? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : lundi 21 novembre 2005 13:34 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Can not build zaptel with kernel-2.6.12 Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can not build zaptel with kernel-2.6.12
Hello David, I rewrote the Makefile so I can compile the modules . However I got the same problems with kernel 2.4.I fixed some variables which was not found . Is it a problem with my debian installation !!??? Regards Harry PS: I like to set ! for Mr Pascal :-) --- David Uzzell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // I have to ask the obvious question. Do you have the same source as you have kernel running? Remember if you have run an upgrade it could have updated the kernel but may not have doen the sources and if you have the sources from the installion media then you would have different versions that will cause this exact problem. David Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Andrew Kohlsmith [EMAIL PROTECTED] wrote: It doesn't really matter whether you buy it (my explanation) or not -- if your specific echo is greater than what the software and/or hardware are designed to handle, it will work poorly. It's called a misapplication of the technology. Two products are both intended to eliminate echo, and product A, due to it's design, can't eliminate some of the echos that product B can. It seems quite fair to say that B is a better product than A. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Broadvoice
Folks, I have several SIP providers that work fine. But I just added a Broadvoice account and all I seem to get is Your call cannot be completed at this time. Broadvoice is registered and receives incoming calls. Dial plan (identical to other external SIP providers) is passing call to Broadvoice. Broadvoice tech surprise is no help (no surprise). They are telling me to add a 0 or 9 to the dial plan, but I see nothing in their documentation to suggest I need to append this 0 or 9 so I have no idea what they are talking about. asterisk -r -- Called broadvoice/15202030583 -- Started music on hold, class 'default', on channel 'SIP/david-f08e' -- SIP/broadvoice-3556 is ringing -- SIP/broadvoice-3556 answered SIP/david-f08e -- Stopped music on hold on SIP/david-f08e -- Attempting native bridge of SIP/david-f08e and SIP/broadvoice-3556 == Spawn extension (longdistance3, 15202030583, 1) exited non-zero on 'SIP/david-f08e' Note that my asterisk server is _not_ showing me congestion, so the message must be coming from Broadvoice. Note: yes, I have user=phone in sip.conf for Broadvoice. TIA, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Death at 2am
In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote: Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. I did look at that. Nothing seems to run at 2am that doesn't run on every other hour. My first inclination was that maybe newsyslog was trying to rotate Asterisk's logs at 2am, but that doesn't look to be the case. I take it that Asterisk (with mostly the standard sample config files) doesn't try to do anything at 02:00 then? I don't believe it does anything according to a fixed schedule like that. Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. I hadn't considered that. Connectivity provider, or VOIP provider (of the latter I have more than one)? Could be either. My theory is that something your Asterisk tries to do regularly is failing for some reason at that time, due to something either internal or external, and that the error handling is not closing one or more file descriptors that it had opened. As the failed operation gets retried (possibly quickly and often), these leaked fd's accumulate and eventually reach the process limit. You could also add or uncomment the following line in /etc/asterisk/logger.conf: full = notice,warning,error,debug,verbose (you may have to restart asterisk or do a logger reload) Then you can see what is in /var/log/asterisk/full after the problem has occurred. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stop asterisk when Idle
This are the facts: after a couple of days running, everything appears to run very well.. asterisk is alive, no bad lines in log.. But actuallu th oh323 channel disappears if tou type at the console oh323 TAB no helps is given oh323: no such command !!! help: nothiong about oh323 !!! but the box is the same as ine hour before, when oh323 was known I am not an asterisk programmer, I am sorry i never read a line of the oh323 channel, so 99.7 % of what I say is wrong, but it seem to me then the oh323 crashes in such a bad way to completly evanihes. If you have running conversation, they stay up (of course, now they are trunked to a zap channel) but no new conversation are possible the really incredible thing is that you cannot find any proble/error line both in the /vat/log/asterisk/full than in the messages log So, that's way I should stop and reboot the box If anybody has any idea (where to look i.e.,...) I can try almost anything Remember, I have a large number of calls to handle (more the 10,000 in a day) Andrea snacktime [EMAIL PROTECTED] com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 17/11/2005 23.30 Re: [Asterisk-Users] stop asterisk when Idle Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com But I found some situations that, after several millions of calls seconds, need to reboot the box and not only restart asterisk. That's really not necessary,and it's almost painful to watch people do this... If you posted some detailed information about your system and the problem you are having maybe someone could help you fix the actual problem. Chris___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstLinux 0.2.9 Released
Hello Everyone, I have finished up work on what will (hopefully) become AstLinux 0.3.0. AstLinux 0.2.9 has been released as a test release, and includes the following changes: - Asterisk 1.2.0 - Zaptel 1.2.0 - libpri 1.2.0 - Sangoma wanrouter beta1-2.3.4 - Linux kernel 2.6.13.3 - improved QoS support - dozens of changes brought in from -testing ISO and CF images can be downloaded from http://www.astlinux.org. If everything goes smoothly, 0.3 should be out by mid-week. Thanks! -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HT486 and RFC2833
I've got an HT486 on my network which was configured to send DTMF over RTP. _Was_ being the operative word because post my recent upgrade to 1.2, RFC2833 for DTMF just stopped working! Works fine on my GXP2000, but no longer on the HT486. Got it going again by configuring sip info mode. Anyone else had this issue with 1.2 or is it just me? Mark. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] v1.2 and features.conf
Hi, I'm trying to get the 'xfersound' working with v1.2. I enabled all in features.conf (like: xfersound = beep), but I can't get the beep when transferring a call. I'm trying this with * v1.2, the bristuff-version, but I'm not sure if that matters? (does it only work with SIP-to-SIP calls?) Cheers, Kristof. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime Problems
Hi All I have configured asterisk with the addons and setup my config files so that i can pull sip extensions (phones) from a mysql database. I have followed all the docs and have editted my extconfig.conf res_mysql.conf and sip.conf to contain all that is advised. From the CLI i can see realtime has a connection and is able to load the user but when I plug in the voip phone it fails to register. My database content matches that of other phones in the sip.conf. If i remove the database user and add it direct into sip.conf the phone connects fine. Any help would be appreciated as its driving me mad now Very Happy CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username scott for 22 minutes, 47 seconds. CLI realtime load sipusers name 114 Column Name Column Value id 1 name 114 callerid 114 canreinvite yes context default defaultip 192.168.10.136 dtmfmode info fromuser 114 fullcontact 114 host 192.168.10.136 nat no secret 114 type friend username 114 disallow all allow g729 allow ilbc allow gsm allow ulaw allow alaw regseconds 0 cancallforward yes Nov 21 12:52:55 NOTICE[15585]: chan_sip.c:10793 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.10.136' - Username/auth name mismatch ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP channels
Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote: Hi all, i've a problem in my Asterisk system. We have around 30 SIP phones connected to an asterisk system, and sometimes some SIP channel (associated to an extension) gets busy all the time, even when that extension isn't in use. [..] Any clues? Without any debugging output from your Asterisk server guesses will range from bad SIP phones to bad Asterisk configuration or a small possibility of a bug. When reporting problems like this, you always have to mention which version of Asterisk you are using, which platform and which brand of phone. The more details you deliver, the more likely you will get a good answer that will help you forward. Without any details, you will only get bad answers or answers that will ask you more questions. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstLinux 0.2.9 Released
Hi Kristian, Excellent thanks.. On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote: Hello Everyone, I have finished up work on what will (hopefully) become AstLinux 0.3.0. AstLinux 0.2.9 has been released as a test release, and includes the following changes: - Asterisk 1.2.0 - Zaptel 1.2.0 - libpri 1.2.0 - Sangoma wanrouter beta1-2.3.4 Does this mean the Sangoma S518 ADSL Card may work on Astlinux on a soekris 4810 board do you know? thanks Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crash: using deprecated BYE/Also transfer method
Hi, I'm experiencing some problems with my Asterisk 1.0.9. When a customer tries to use transfer method sometimes Asterisk crashes. The following message appears in /var/log/asterisk/messages Nov 17 15:56:35 WARNING[759]: No path to translate from SIP/12.34.56.78-3aef(1) to SIP/domain.com-b6ccf248(256) Nov 17 15:56:38 NOTICE[759]: Client '12.34.56.78' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead Nov 17 15:56:38 WARNING[759]: Invalid transfer information from '12.34.56.78' But it doesn't crash every time when customer tries to use transfer. Any ideas ? Thanks in advance, Pavel Siderov ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 question
Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New firmware for Aastra/Sayson IP phones
Aastra Telecom has released SIP v1.3 firmware for the Aastra/Sayson range of IP phones. This is a major update compared to firmware 1.2.x with many bugfixes and Asterisk(tm) interop limitations fixed. The firmware, updated manuals and release notes are available for download at: http://www.aastra.com/support/enterpriseip New features include XML scripting support, enhanced integration for Asterisk(tm), Busy Lamp Field (BLF), Multiple SIP Proxy support, HTTP/FTP/TFTP config, encrypted config support, and a complete overhaul of the user and admin documentation. I've posted a quick summary at http://www.voip-info.org/wiki/view/Aastra+480i - email nadlab at aastra.com if you would like a PDF copy of the full spec sheet. ps. Also available is firmware v1.2 for the Aastra CNX, an Asterisk(tm) based conference bridge. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zyxel p2000w
Does anyone know is the zyxel p2000w has call waiting? I hear noise when a second call comes in but cannot find any documentation. Thanks, Chip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Forward Voicemail to remote server?
Hi Are and Matt - We have a user (our CEO) who has phones in two different offices, and we'd like him to be able to get all his VM in either office, regardless of which office was originally called. My idea instead is to use externnotify to run some kind of script to forward the vm to another server. I'm sort of at a loss as to where to start, though. I guess I could rsync the VM files to the other server, and run a name check to rename the VM files if there's a duplicate name. The one problem there are that my coding skills are seriously bad. Of course, what I really want is an asterisk VM system where user accounts are transparent across many servers, and VM's can be shuffled around between the servers as a configuration option in voicemail.conf. I wish I could code and knew Objective-C! Maybe I should submit a feature request. Why don't you use realtime voicemail with the thing for storing messages as blobs? Well that's just plain... Well... Logical ;-) I guess I was being lazy and trying to avoid implementing realtime across all our servers just to satisfy the whims of one user. Since that user does happen to be our CEO, and since it would help realize my dream of transparent users across all our servers, well heck, you've talked me into it. Matt Riddell is right again. (He always is) :-) It does seem that way! Thanks Matt. You just add MySQL replication. It is easy to set up and allows you to have voicemail in as many locations you want. http://dev.mysql.com/doc/refman/5.0/en/replication-howto.html Ahh. I would have been fumbling around trying to do this the wrong way for a long time. Thanks Are! Thanks, Noah Miller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme + sendtext
Yes. The biggest challenge is putting together a mux device that mixes the text frames out to all of the user/channel threads in the conference. On 11/21/05, Olle E. Johansson [EMAIL PROTECTED] wrote: BJ Weschke wrote: On 11/19/05, Jean-Denis Girard [EMAIL PROTECTED] wrote: Hi all, Is sending text to a conference supported by asterisk-1.2, ie one member of the conference sends text, it is received by all other members of the conference (provided their channel supports text of course) ? I made a quick test with IAX softphones, and it seems that text isn't sent to IAX channels through the conference. If it is not supported, how difficult would that be to add this functionnality ? It's not supported at this time, but you're also not the first person to ask about it. I suppose it wouldn't be too difficult to add. Well, we have no routing of incoming text messages through the dial plan, but I guess this is a bit more simple, since we have open channels into meetme. /o ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can anyone explain reason for this echo
On Monday 21 November 2005 07:53, Doug Meredith wrote: Two products are both intended to eliminate echo, and product A, due to it's design, can't eliminate some of the echos that product B can. It seems quite fair to say that B is a better product than A. It depends on your specific needs. If your palate can't tell the difference between a $50 bottle of wine and a $5000 bottle of wine, do you still buy the $5000 bottle because it must be better? As I stated, the software echo can in Zaptel and the hardware echo can from Digium work reasonably well for their intended purpose. If you need something more, then yes, these products are insufficient and you'll need a more powerful (better in your parlance) echo canceller. But for many people, the free one works pretty damn well, so buying a more powerful (better) echo canceller is a waste of money, rack space and power. Yes, I am splitting hairs -- something that solves your problem where something else couldn't doesn't make the latter clearly lacking or complete rubbish across the board. I took exception to your painting the Digium hardware echo can module and the software echo cans in zaptel as trash, as they work very well for many people. They clearly aren't sufficient for your specific needs, and thus the Orion Telecom echo canceller is better -- and I stress this -- for you. It's overkill for many others, and I'm willing to bet that it's still insufficent for others still. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12
Hello Olivier, Non je ne suis pas fâché ! Alors ce *b2bua ? En fait je cherche une solution pour intègrer SER+Asterisk sur la même machine. Ser est un bon proxy asterisk un bon ipbx. Je souhaite utilisé ser pour le routage sip avec asterisk et pour fournir les service de téléponie d'entreprise plus l'IM et presence via SIMPLE qu'asterisk ne propose pas ! Mon problème est le champ contact dans le Sip HF avec des clients natés Une idée ? Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Salut Harry, plus de nouvelles de toi :( Serais tu faché? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : lundi 21 novembre 2005 13:34 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Can not build zaptel with kernel-2.6.12 Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problems
Did you see the mysql.log file? I was having a similar problem, and i saw a problem with an update in a mysql table when a user was trying to register a phone. Sixto - Original Message - From: scott [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, November 21, 2005 4:24 AM Subject: [Asterisk-Users] Realtime Problems Hi All I have configured asterisk with the addons and setup my config files so that i can pull sip extensions (phones) from a mysql database. I have followed all the docs and have editted my extconfig.conf res_mysql.conf and sip.conf to contain all that is advised. From the CLI i can see realtime has a connection and is able to load the user but when I plug in the voip phone it fails to register. My database content matches that of other phones in the sip.conf. If i remove the database user and add it direct into sip.conf the phone connects fine. Any help would be appreciated as its driving me mad now Very Happy CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username scott for 22 minutes, 47 seconds. CLI realtime load sipusers name 114 Column Name Column Value id 1 name 114 callerid 114 canreinvite yes context default defaultip 192.168.10.136 dtmfmode info fromuser 114 fullcontact 114 host 192.168.10.136 nat no secret 114 type friend username 114 disallow all allow g729 allow ilbc allow gsm allow ulaw allow alaw regseconds 0 cancallforward yes Nov 21 12:52:55 NOTICE[15585]: chan_sip.c:10793 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.10.136' - Username/auth name mismatch ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crash: using deprecated BYE/Also transfer method
Pavel Siderov wrote: Hi, I'm experiencing some problems with my Asterisk 1.0.9. When a customer tries to use transfer method sometimes Asterisk crashes. The following message appears in /var/log/asterisk/messages Nov 17 15:56:35 WARNING[759]: No path to translate from SIP/12.34.56.78-3aef(1) to SIP/domain.com-b6ccf248(256) Nov 17 15:56:38 NOTICE[759]: Client '12.34.56.78' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead Nov 17 15:56:38 WARNING[759]: Invalid transfer information from '12.34.56.78' But it doesn't crash every time when customer tries to use transfer. Any ideas ? Again, without full information it's very hard to diagnose your problem. Please turn on debug to 4, verbose to 4 and turn on SIP debug and capture all the traffic from one of these failed transactions. Attach those in a bug report in bugs.digium.com. Also, please test with the Asterisk 1.2 release version, even though I belive that nothing has changed in the old Bye/Also scheme. Please also tell us what phones you use. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip show users
asterisk1*CLI sip show users UsernameSecret Accountcode Def.Context ACL NAT 205 testfrom-internal No No 204 testfrom-internal No No 203 testfrom-internal No No 202 020 from-internal No No 201 testfrom-internal No No how can I get this information in my asterisk Macros in extesions.conf something like sip(204(Def.Context))=from-internal ?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MySQL - Realtime install procedure?
I'd like to begin messing around with realtime and mysql, but have never done anything with either before. Can anyone point me to any form of document that would help me understand the installation/config process? Been around * for a couple of years and linux for more then ten years, just never messed with mysql, etc. I'm certainly a newb with this piece. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can anyone explain reason for this echo
Andrew Kohlsmith [EMAIL PROTECTED] wrote: I took exception to your painting the Digium hardware echo can module and the software echo cans in zaptel as trash, as they work very well for many people. They clearly aren't sufficient for your specific needs, and thus the Orion Telecom echo canceller is better -- and I stress this -- for you. That wasn't me. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] split line authorization problem (ATL IP400 phone)
Hi, I'm using an ATL IP400 phone and cant get it to register, it fails with: chan_sip.c:9405 handle_request_register: Registration from 'xx sip:[EMAIL PROTECTED]' failed for 'x.x.x.x' looking at the register request i notice two things: Authorization: Digest username=xx,realm=telecomplete,nonce=30c9c7a4,uri=sip:yy, response=d42ad1c6bb1ccce2374e8fef69a92704 it does not specify algorithm=md5 and the Authorization line is split onto two lines. Looking at the code, it doesnt seem to check for the algorithm field so i'm thinking that part is okay, but i cant confirm if it is parsing the SIP headers and joining up the split lines before it carries out check_auth. any ideas? thanks Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Can anyone explain reason for this echo
On Monday 21 November 2005 09:23, Doug Meredith wrote: That wasn't me. hahaha you're quite right. I wasn't paying attention to who replied. My apologies. I feel that my points still apply, though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] addmailbox script
Hello, I checked out the asterisk version from the CVS. But I dont seem to have the addmailbox script. How can I setup a mailbox without this utility. Regards, Rajesh Golani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please Help with Zaptel
Can someone tell me what problem I am having with Zaptel on a Suse 10 distribution? 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 ./makefw pciradio.rbt radfw radfw.h Loaded 42096 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 libtonezone.a -lm 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 -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.13-15-default/build make -C /lib/modules/2.6.13-15-default/build SUBDIRS=/root/zaptel-1.2.0 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15-obj/i386/default' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.13-15-obj/i386/default' make: *** [linux26] Error 2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL - Realtime install procedure?
Try out http://astbill.com AstBill is an Open Source Web Based Billing, Routing and Management Software for Asterisk and MYSQL. It is using 100% Realtime and there is active support in the forum. http://astbill.com/forum-- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIPAstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problems
You have error: Username/auth name mismatch So there is clearly and issue with the content in your table. In our setup the column name and username have the same value = 114 the fromuser and authuser column = NULL If this is not helping send your table definition and the content of your record 114 and we will sort it out. -- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultantshttp://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP installation
How do you install AMP? I downloaded it and tried to run make or install and it doesnt work. Is there some trick to this? Thank.s ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk versions after the 1.2 release
Friends in the Asterisk community, There have been a lot of questions about Asterisk version numbers on the mailing lists. Here's a clarification: * Executive summary --- - Asterisk 1.2 = RELEASE version (previously called stable) Asterisk 1.2.0 = First release of 1.2 (released now) Asterisk 1.2.1 = Second release of 1.2 (not out yet) - Asterisk 1.0 = old version, not maintained any more Asterisk 1.0.9 = Final release of Asterisk 1.0 - Asterisk 1.3 = DEVELOPMENT code base, dangerous territory * Asterisk 1.2 is the RELEASE version - This version is maintained in the v1-2 CVS branch. The released code, that you want to use in production servers, is released as tar.gz archives on ftp.digium.com and mirrors. These reflect the tagged CVS code, the first tag in the 1.2 tree being v1-2-0. The next release will be version 1.2.1, consisting of updated code including bug fixes that has been done since the 1.2 release date. From the minute of the release, we've had a lot of interest in testing the new version and a constant flow of bug reports. You do not want to follow the v1-2 CVS tree in production, since it is changing quite a lot and not all changes are tested and stabilized. After testing, we product tar.gz archives that you want to use. Make sure you subscribe to the asterisk-announce mailing list to get updates if you do not follow the massive flow of messages in asterisk-users. For the 1.2 tree functionality is frozen. No new functionality is added to this code. The rule is that we only apply additional documentation and bug fixes to a release version. * Asterisk 1.0 is no longer maintained -- The old release version, 1.0 is no longer maintained, apart from the possibility of serious security bugs that needs to be fixed. This code is over one year old now and we've successfully managed to avoid adding new functionality to it since the release in september 2004. Before filing a bug report for a 1.0 version, make sure you also test the 1.2 version - a lot of things have been fixed in 1.2. If the bug exists in 1.2, go ahead and make a note in the bug report that it doesn't work in either version. * Asterisk 1.3 is the new development code base --- CVS head is the name currently used for the development code base, which now is on version 1.3dev. This is the base for a future 1.4 RELEASE. During the development process of 1.3, we will move from CVS as a versioning system to subversion, a new system used by many open source projects. cvs head will not be a useful name for 1.3dev for much longer. WARNING :: Be warned that developers will go crazy with this code after a long period of bug-fixing and release engineering with 1.2. New things will be added quickly, and this version may or may not work at all from time to time. A lot of quite large internal architectures changes will be implemented in 1.3. These will have to be tested and propably cause a lot of very interesting craches, from a coding perspective. From a user perspective, those crashes will not be interesting or fun at all. Avoid the development tree in production use. I hope this message clarifies the confusion a bit. Regards, /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problems
Hi Thank you for your reply. I have tried various definitions in the sipusers table but none seem to be working :-( I have attached mey structure and content export below for your attention. Many thanks Scott Pinhorne -- -- Table structure for table `sip_users` -- CREATE TABLE `sip_users` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `fullcontact` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` varchar(5) NOT NULL default 'no', `deny` varchar(95) default NULL, `permit` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 ROW_FORMAT=DYNAMIC AUTO_INCREMENT=2 ; -- -- Dumping data for table `sip_users` -- INSERT INTO `sip_users` VALUES (1, '114', NULL, NULL, NULL, '114', 'yes', 'default', '192.168.10.136', 'info', NULL, NULL, '114', '192.168.10.136', NULL, NULL, NULL, NULL, 'no', NULL, NULL, NULL, NULL, '', NULL, NULL, NULL, NULL, '114', 'friend', '114', 'all', 'g729;ilbc;gsm;ulaw;alaw', NULL, 0, '', '', 'yes'); -Original message- From: Are [EMAIL PROTECTED] Date: Mon, 21 Nov 2005 09:07:43 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Realtime Problems You have error: Username/auth name mismatchhttp://fast.turbosite.net/phpmyadmin/tbl_properties_structure.php? lang=en-utf-8server=1collation_connection=utf8_general_cidb=mans ionpbxtable=asv_sip So there is clearly and issue with the content in your table. In our setup the column name and username have the same value = 114 the fromuser and authuser column = NULL If this is not helping send your table definition and the content of your record 114 and we will sort it out. -- Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please Help with Zaptel
It looks like you do not have the kernel source code installed. Go to 'Yast' and 'Install Software'. Look for the package called 'kernel-source'. It will install the source for your kernel. Then run the 'Update Software' to make sure the kernel and the kernel source are the same version. Then try compiling again. Dan /lib/modules/2.6.13-15-default/build make -C /lib/modules/2.6.13-15-default/build SUBDIRS=/root/zaptel-1.2.0 modules make[1]: Entering directory `/usr/src/linux-2.6.13-15-obj/i386/default' make[1]: *** No rule to make target `modules'. Stop. make[1]: Leaving directory `/usr/src/linux-2.6.13-15-obj/i386/default' make: *** [linux26] Error 2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not picking up calls.
Hiya, anyone have an idea what I need to do to fix this, I have a TDM400P and asterisk 1.2, when I make a call to the system asterisk see the phone ringing and looks like it picks it up from the console, but the phone actually just continues to ring. I am thinking I have something silly in the config or the cable from the TDM400P to the phone socket is dodgy. Anyone got any thoughts? Mark -- Starting simple switch on 'Zap/4-1' Nov 21 15:15:27 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:28 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:30 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Nov 21 15:15:31 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... Nov 21 15:15:33 NOTICE[13716]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... -- Detected ring pattern: 386,366,266 -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Set(Zap/4-1, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing Set(Zap/4-1, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(Zap/4-1, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') Nov 21 15:15:36 WARNING[13716]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 localhost*CLI /etc/zaptel.conf loadzone=uk defaultzone=uk fxsks=4 /etc/asterisk/zapata.conf [channels] language=en callwaiting=no callprogress=no usecallerid=yes echocancel=yes echocancelwhenbridged=no usedistinctiveringdetection=yes busydetect=no rxgain=0.0 txgain=0.0 group=1 immediate=no context=inbound-from-pstn signalling=fxs_ks channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi You can give me some idea of as do it. Actaually I've the following trial network endpoint -- GK1 -- GK2 -- Asterisk GK1 configuration: Direct Mode GK2 configuration : Routed Mode Thanks in advance!! Best Regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk versions after the 1.2 release
Hello, Several of us were told that there would be a 1.0.10 release as the final release of Asterisk 1.0 tree. There are several serious bugs in the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to have this packaged as a release before the tree stops being accessible on the CVS server. If all that is needed is someone to do it, I volunteer to tar-gz it up and call it a release. Yes, I am using 1.2 on several Production servers right now, but I do see the need for there to be a final release of 1.0 before it is buried for good. Thanks, MATT--- On 11/21/05, Olle E. Johansson [EMAIL PROTECTED] wrote: Friends in the Asterisk community, There have been a lot of questions about Asterisk version numbers on the mailing lists. Here's a clarification: * Executive summary --- - Asterisk 1.2 = RELEASE version (previously called stable) Asterisk 1.2.0 = First release of 1.2 (released now) Asterisk 1.2.1 = Second release of 1.2 (not out yet) - Asterisk 1.0 = old version, not maintained any more Asterisk 1.0.9 = Final release of Asterisk 1.0 - Asterisk 1.3 = DEVELOPMENT code base, dangerous territory * Asterisk 1.2 is the RELEASE version - This version is maintained in the v1-2 CVS branch. The released code, that you want to use in production servers, is released as tar.gz archives on ftp.digium.com and mirrors. These reflect the tagged CVS code, the first tag in the 1.2 tree being v1-2-0. The next release will be version 1.2.1, consisting of updated code including bug fixes that has been done since the 1.2 release date. From the minute of the release, we've had a lot of interest in testing the new version and a constant flow of bug reports. You do not want to follow the v1-2 CVS tree in production, since it is changing quite a lot and not all changes are tested and stabilized. After testing, we product tar.gz archives that you want to use. Make sure you subscribe to the asterisk-announce mailing list to get updates if you do not follow the massive flow of messages in asterisk-users. For the 1.2 tree functionality is frozen. No new functionality is added to this code. The rule is that we only apply additional documentation and bug fixes to a release version. * Asterisk 1.0 is no longer maintained -- The old release version, 1.0 is no longer maintained, apart from the possibility of serious security bugs that needs to be fixed. This code is over one year old now and we've successfully managed to avoid adding new functionality to it since the release in september 2004. Before filing a bug report for a 1.0 version, make sure you also test the 1.2 version - a lot of things have been fixed in 1.2. If the bug exists in 1.2, go ahead and make a note in the bug report that it doesn't work in either version. * Asterisk 1.3 is the new development code base --- CVS head is the name currently used for the development code base, which now is on version 1.3dev. This is the base for a future 1.4 RELEASE. During the development process of 1.3, we will move from CVS as a versioning system to subversion, a new system used by many open source projects. cvs head will not be a useful name for 1.3dev for much longer. WARNING :: Be warned that developers will go crazy with this code after a long period of bug-fixing and release engineering with 1.2. New things will be added quickly, and this version may or may not work at all from time to time. A lot of quite large internal architectures changes will be implemented in 1.3. These will have to be tested and propably cause a lot of very interesting craches, from a coding perspective. From a user perspective, those crashes will not be interesting or fun at all. Avoid the development tree in production use. I hope this message clarifies the confusion a bit. Regards, /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: AMP installation
On Monday 21 November 2005 17:12, Goran Donev wrote: How do you install AMP? I downloaded it and tried to run make or install and it doesn't work. Is there some trick to this? The trick is to run the install script and read the documentation. Just not in that order... -- Cheers Wayne ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Contact field in SIP HF between asterisk + ser
___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! Hello, As far as I know Asterisk cannot disentangle RTP from signaling in either SIP or H323 at least until now. I'd also be interested to know if this option is available now in case I've missed something... Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: SIP firmware image for Cisco 7940 or 7960
You can download a new SIP firmware and force the Cisco IP phone to use it. Some interesting links about it: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Joao Daryl Johnson wrote: Sorry for the off topic message, but I am ready to give up on this 7940... I don't know what firmware version is loaded, but based on the sniffer traces it appears to be SIP 5.x or better... The problem is that I don't have any firmware files for this device. Can anyone point me in the right direction? Thanks for the help, Daryl ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] addmailbox script
On Mon, Nov 21, 2005 at 08:18:09PM +0530, Rajesh Golani wrote: Hello, I checked out the asterisk version from the CVS. But I dont seem to have the addmailbox script. Because it is no longer needed How can I setup a mailbox without this utility. app_voicemail does that for you. No need to bother. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Gateway Providers
VoicePulse IAX.cc BroadVoice Teliax Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com Hi, Can anyone recommend a good reputable VoIP gateway service provider that I can use with my Asterisk server in wa.us? All I can seem to find is VoIP service directly to the desk. I'd prefer a provider that can provide DID-type services, because that is my big selling point to the company. Thanks, Jeff Ramsey MIS Administrator Tubafor Mill, Inc. [EMAIL PROTECTED] 360.269.1650 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.4/176 - Release Date: 11/20/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crash: using deprecated BYE/Also transfer method
Hi,
It's not possible to provide log due to the reason that system is in
production and there are many current calls. Crash happens on 1-2 weeks
once. I cannot simulate and get the same result with x-lite, cisco ata
and sipura 3000 when trying transfer. But some of the customers some
way crash asterisk. I don't know what UACs do they use. The dial string
I'm using is - exten = _00.,1,Dial,SIP/${[EMAIL PROTECTED]
Thanks,
Pavel
Olle E. Johansson wrote:
Pavel Siderov wrote:
Hi,
I'm experiencing some problems with my Asterisk 1.0.9. When a customer
tries to use transfer method sometimes Asterisk crashes. The following
message appears in /var/log/asterisk/messages
Nov 17 15:56:35 WARNING[759]: No path to translate from
SIP/12.34.56.78-3aef(1) to SIP/domain.com-b6ccf248(256)
Nov 17 15:56:38 NOTICE[759]: Client '12.34.56.78' using deprecated
BYE/Also transfer method. Ask vendor to support REFER instead
Nov 17 15:56:38 WARNING[759]: Invalid transfer information from
'12.34.56.78'
But it doesn't crash every time when customer tries to use transfer.
Any ideas ?
Again, without full information it's very hard to diagnose your problem.
Please turn on debug to 4, verbose to 4 and turn on SIP debug and
capture all the traffic from one of these failed transactions. Attach
those in a bug report in bugs.digium.com.
Also, please test with the Asterisk 1.2 release version, even though I
belive that nothing has changed in the old Bye/Also scheme.
Please also tell us what phones you use.
/Olle
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Re: [Asterisk-Users] AstLinux 0.2.9 Released
this would be very beneficial to me as well.. I have the S518 ADSL card in my Linux system as well.. I was looking at going to an ASTLINUX solution. Hi Kristian, Excellent thanks.. On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote: Hello Everyone, I have finished up work on what will (hopefully) become AstLinux 0.3.0. AstLinux 0.2.9 has been released as a test release, and includes the following changes: - Asterisk 1.2.0 - Zaptel 1.2.0 - libpri 1.2.0 - Sangoma wanrouter beta1-2.3.4 Does this mean the Sangoma S518 ADSL Card may work on Astlinux on a soekris 4810 board do you know? thanks Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk versions after the 1.2 release
Matt Florell wrote: Hello, Several of us were told that there would be a 1.0.10 release as the final release of Asterisk 1.0 tree. There are several serious bugs in the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to have this packaged as a release before the tree stops being accessible on the CVS server. I haven't heard of that promise, I'll ask Russel about it. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_bluetooth and Audiovox 6600 problem
Hello. I have sucessfully installed chan_bluetooth with my asterisk system. However I wasn't able to get to that until I completed a few other steps.. 1) using the sdptool - start up the services that the Audiovox is looking to pair with 'sdptool hs' 'sdptool hf' this allowed me to start the pairing with an audio gateway on my vx 6600. However, when the chan_bluetooth tries to initialize the channel i get nothing but errors on the console back from the audiovox. Are there specific strings that are needed to send to initialize individual phones? When I do a bluetooth show peers, it shows that it is Ready, but no signal. Yet, on the VX6600 it shows signal at just right on it's meter bar. The three values are (too weak, just right, too strong). Where should I look next? Audiovox's implmentation of Bluetooth? Anyone ?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crash: using deprecated BYE/Also transfer method
Pavel Siderov wrote: Hi, It's not possible to provide log due to the reason that system is in production and there are many current calls. Crash happens on 1-2 weeks once. I cannot simulate and get the same result with x-lite, cisco ata and sipura 3000 when trying transfer. But some of the customers some way crash asterisk. I don't know what UACs do they use. The dial string I'm using is - exten = _00.,1,Dial,SIP/[EMAIL PROTECTED] Well, until we know what phone they use so we can repeat it, or you can get a log file, there's nothing much we can do about it. Sorry. It must be a pretty old firmware or version of a softphone to use BYE/ALSO. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bluetooth headset with softphone or direct asterisk
I'll try that tonight... On Sat, 2005-11-19 at 13:47 -0800, Ben Higley wrote: [AG] Pocket_PC AT+BRSF=23 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CIND=? [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CIND? [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CMER=3,0,0,1 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CLIP=1 [AG] Pocket_PC ERROR [AG] Pocket_PC AT+CGMI=? [AG] Pocket_PC ERROR Strange behaviour. Do you get similar behaviour if you connect to the phone with minicom? Are you connecting on the correct channel? Show sdptool browse output for the phone. -- dwmw2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone parked in your Asterisk?
Based on a discussion on the IRC a long time ago (several days) I've created a patch for 1.2 in the bug tracker that allows you to see if a parking lot is occupied or not - provided you use the Flash panel or SIP subscriptions. What you do: * Patch the 1.2 source with the patch in http://bugs.digium.com/view.php?id=5779 * Add an extension in your dialplan with a hint that uses the local channel (yes, the patch adds device state to the local channel) exten= 100,hint,local/[EMAIL PROTECTED] * Add a subscription to this extension in a SIP phone, like Eye-beam * As soon as there's a call parked on that parking log (701), you will see it visually - how depends on the phone. This hack possibly have other uses as well, feel free to explore. If it doesn't work, tell me more in the bug tracker. If it does work, tell me about it in the bug tracker. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration Problem
I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server: Transmitting (no NAT) to 10.1.1.152:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 10.1.1.152;branch=z9hG4bK4b554363d4497847;received= 10.1.1.152From: sip:[EMAIL PROTECTED];tag=12e8dd0080754148To: sip:[EMAIL PROTECTED];tag=as2383b1dfCall-ID: [EMAIL PROTECTED]CSeq: 100 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=54cb5290Content-Length: 0 After a few server restarts and/or phone restarts the phone registers ok. Any ideas why ? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA941
Just picked up two of these puppies from my parcelforce depot. Man, they are smart phones. They look the business. I installed one within seconds, fantastic web configuration - much like the SPA3000 box. Speakerphone sounds good, handset feels and sounds good. I'll be using this heavily over the next couple of days, and I'll let you all know how we find it. And nearly half the price of a second-hand 7940 it's a real steal. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you disable realtime?
Am I correct in assuming that if I am not running Realtime on my asterisk 1.2 server, the proper way to disable it is to remove the following 2 files: /usr/lib/asterisk/modules/pbx_realtime.so /usr/lib/asterisk/modules/app_realtime.so I am just testing out the default installation and am getting these errors on the console: Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Any help will be appreciated. - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone parked in your Asterisk?
Does it hold state information for any channel? Even ZAP, IAX, etc!!! If it does, Olle, you have just placed us one step closer to being able to emulate a Key system!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, November 21, 2005 11:45 AM To: Users Asterisk Subject: [Asterisk-Users] Anyone parked in your Asterisk? Based on a discussion on the IRC a long time ago (several days) I've created a patch for 1.2 in the bug tracker that allows you to see if a parking lot is occupied or not - provided you use the Flash panel or SIP subscriptions. What you do: * Patch the 1.2 source with the patch in http://bugs.digium.com/view.php?id=5779 * Add an extension in your dialplan with a hint that uses the local channel (yes, the patch adds device state to the local channel) exten= 100,hint,local/[EMAIL PROTECTED] * Add a subscription to this extension in a SIP phone, like Eye-beam * As soon as there's a call parked on that parking log (701), you will see it visually - how depends on the phone. This hack possibly have other uses as well, feel free to explore. If it doesn't work, tell me more in the bug tracker. If it does work, tell me about it in the bug tracker. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer and pick chan_h323
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk
1.0.x dtmf=inband registers and sends dtmf but with asterisk 1.2 and
ooh323 it does not.. is this a known issue ? While google heard tht
there was a issue with chan_h323.so would not send inband so tried to
install chan_0h323.so but but.. asterisk refuses to start with
chan_oh323 it says Unregistered channel type 'Modem'
my basic requirements are h323 , call pickup and call transfer? below
attached are the configurations files tht we are using currently ...
thanking for all your support ..
Extensions.conf:-
[testing]
exten = _7.,1,Pickup({66}:[EMAIL PROTECTED])
exten = 666,1,Dial(H323/192.168.1.194,100,Ttr)
exten = 667,1,Dial(H323/192.168.1.195,100,Ttr)
exten = 668,1,Dial(H323/192.168.1.196,100,Ttr)
exten = 669,1,Dial(H323/192.168.1.192,100,Ttr)
H323.conf:-
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for
this machine
disallow=all
allow=ulaw
allow=alaw
;dtmfmode=auto
dtmfmode=inband
gatekeeper = DISABLE
context=testing
[vivek]
type=friend
host=192.168.1.194
context=testing
Callgroup=1
pickupgroup=1-9,13
[santosh]
type=friend
host=192.168.1.195
context=testing
Callgroup=1
pickupgroup=1-9,13
[binu]
type=friend
host=192.168.1.196
context=testing
Callgroup=1
pickupgroup=1-9,13
[test1]
type=friend
host=192.168.1.192
context=testing
Callgroup=1
pickupgroup=1-9,13
Features.conf:-
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
pickupex = *8
[featuremap]
blindxfer = #1 ; Blind transfer
atxfer = *2 ; Attended transfer
I haven't lost my mind; it's backed up on
tape somewhere.
Santosh Rao
Trikon Electronics Pvt Ltd
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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RE: [Asterisk-Users] Linksys SPA941
We have a review of it at http://voipspeak.net, I personally really like it. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Monday, November 21, 2005 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Linksys SPA941 Just picked up two of these puppies from my parcelforce depot. Man, they are smart phones. They look the business. I installed one within seconds, fantastic web configuration - much like the SPA3000 box. Speakerphone sounds good, handset feels and sounds good. I'll be using this heavily over the next couple of days, and I'll let you all know how we find it. And nearly half the price of a second-hand 7940 it's a real steal. Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.4/176 - Release Date: 11/20/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.4/176 - Release Date: 11/20/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do you disable realtime?
It is a better practice to use a noload option in modules.conf. That way if and when you upgrade you wont need to remove them again they will just continue to not load Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PedroSent: Monday, November 21, 2005 12:11 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] How do you disable realtime? Am I correct in assuming that if I am not running Realtime on my asterisk 1.2 server, the proper way to disable it is to remove the following 2 files:/usr/lib/asterisk/modules/pbx_realtime.so/usr/lib/asterisk/modules/app_realtime.soI am just testing out the default installation and am getting these errors on the console:Nov 21 12:05:29 ERROR[30656] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info.Nov 21 12:05:29 WARNING[30656] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug.Any help will be appreciated.- Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_icd anyone? on 1.2?
Well, this is interesting - is anybody actually using app_icd out there? :-) l. On Thu, 17 Nov 2005 00:54:56 +0100, Tyler [EMAIL PROTECTED] wrote: Anyone using app_icd? I need to use some of the advanced features that the regular asterisk Queue() application won't provide. Anyone have any configuration examples, etc? Will it work with the current 1.2rc release? Thanks tf. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone parked in your Asterisk?
On 11/21/05, Alexander Lopez [EMAIL PROTECTED] wrote: Does it hold state information for any channel? Even ZAP, IAX, etc!!! If it does, Olle, you have just placed us one step closer to being able to emulate a Key system!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, November 21, 2005 11:45 AM To: Users Asterisk Subject: [Asterisk-Users] Anyone parked in your Asterisk? Based on a discussion on the IRC a long time ago (several days) I've created a patch for 1.2 in the bug tracker that allows you to see if a parking lot is occupied or not - provided you use the Flash panel or SIP subscriptions. What you do: * Patch the 1.2 source with the patch in http://bugs.digium.com/view.php?id=5779 * Add an extension in your dialplan with a hint that uses the local channel (yes, the patch adds device state to the local channel) exten= 100,hint,local/[EMAIL PROTECTED] * Add a subscription to this extension in a SIP phone, like Eye-beam * As soon as there's a call parked on that parking log (701), you will see it visually - how depends on the phone. This hack possibly have other uses as well, feel free to explore. If it doesn't work, tell me more in the bug tracker. If it does work, tell me about it in the bug tracker. I believe it holds state information for parked channels by using the Local channel to get at the status of the parked extensions. It's a very nice new feature, no doubt. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk crash: using deprecated BYE/Also transfer method
Could you please advice me how to create log all calls or only for
those using Bye/Also. I've made some researche using google and found
that SJPhone use this method -
http://www.sjlabs.com/doc/SJphone%20Profiles.pdf .
Thanks in advance,
Pavel
Olle E. Johansson wrote:
Pavel Siderov wrote:
Hi,
It's not possible to provide log due to the reason that system is in
production and there are many current calls. Crash happens on 1-2 weeks
once. I cannot simulate and get the same result with x-lite, cisco ata
and sipura 3000 when trying transfer. But some of the customers some way
crash asterisk. I don't know what UACs do they use. The dial string I'm
using is - exten = _00.,1,Dial,SIP/${[EMAIL PROTECTED]
Well, until we know what phone they use so we can repeat it, or you can
get a log file, there's nothing much we can do about it. Sorry.
It must be a pretty old firmware or version of a softphone to use BYE/ALSO.
/O
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[Asterisk-Users] How to deal with echo in MeetMe?
I have a customer who is running fairly large conferences (between 5 and 30 participants) on their Asterisk box. It uses SIP to talk to a PSTN provider. They are complaining that under some circumstances they experience echo of one or more participants. On listening in to one of their conferences, it seemed to me that the echo was being introduced via the microphone of a couple of specific participants, as it was possible to eliminate this echo by muting those participants. On discussing the participants' environments with the customer, it would appear that the problem occurs when participants are using speaker phones and there are multiple participants in proximity to each other, such that one participant's phone can hear the audio from that of another participant in the same conference. It's my supposition that any echo canceller is going to have difficulties correcting for that scenario. Am I correct? The problem I have is that the customer insists that their existing conferencing supplier (whom our kit is supposed to replace) does not suffer from this echo, in the same participant environment. I am assured by our PSTN supplier that there is full echo suppression on the PSTN lines. Am I correct in believing that further echo suppression is neither possible nor required at the SIP interface within Asterisk? Any advice on how to approach this would be appreciated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
