[asterisk-users] SIP_CODEC related question
hello ALL, My question is regarding SIP_CODEC. 1). How can I get which codec is used for this channel . Ex: if incoming call to asterisk i want to know which codec is used for this channel. is there any way for printing codec in dial plan 2). How can I set codec for outbound dialing. ex: In 1.0 there is some variable called SIP_CODEC which can be setted . what about newer version like 1.6 or greater. is anybody know regarding this regards Dhaval.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to backup Trixbox 2.8.0.3
Hi all, I have installed Trixbox 2.8.0.3 with TDM400p digium cards (2 fxs - 2 fxo). Before making security settings I need to backup all system. When I click on Backup on System - Backup menu in admin panel anything come up on screen. I tried to observe another way some users suggested mondo for backup. With mondo I had had some problems before installation. Can anyone tell me how can we backup completely Trixbox with effective way without problems ? Thanks Yavuzhan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
On Wed, Dec 09, 2009 at 07:57:52AM +0100, Olivier wrote: What's the output of: lspci -v -nn -s 08:00.0 # lspci -v -nn -s 08:00.0 08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] [1397:16b8] (rev 01) Subsystem: Cologne Chip Designs GmbH Device [1397:b552] Flags: medium devsel, IRQ 10 I/O ports at ec00 [size=8] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hfcmulti, wcb4xxp Hmm... you're right. This seems to be a typo in PCI.pm. Fixed, thanks. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SkypeForAsterisk
Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in advance Srinivas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_voicemail. Help me to find typo source ...
On Wed, Dec 09, 2009 at 08:54:13AM +0100, Olivier wrote: Hi, In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see : [Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing '/var/spool/asterisk/voicemail/default/103/INBOX/msg.txt': [Dec 8 15:02:17] VERBOSE[10283] config.c: == Found [Dec 8 15:02:17] VERBOSE[10283] file.c: -- SIP/103-0784 Playing 'vm-message.gsm' (language 'fr') [Dec 8 15:02:18] WARNING[10283] file.c: File vm-recieved does not exist in any format [Dec 8 15:02:18] WARNING[10283] file.c: Unable to open vm-recieved (format 0x8 (alaw)): No such file or directory [Dec 8 15:02:18] WARNING[10283] say.c: Unable to play message vm-recieved Strangely, I can't find any line in *.c files containing such vm-recieved string. If I add ln -s vm-received vm-recieved in appropriate directory, those error messages disappear. Though I've found a work around, I would be very happy to pin point root cause but I'm a bit lost in the amount of files in Asterisk code. Interesting. Grepping for 'recieve' in 1.6.2.0-rc6 gives only two hits: 1. in the Changelog: 009-07-27 20:33 + [r209234] David Brooks dbro...@digium.com * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces recieved with received and initilize with initialize (closes issue #15571) Reported by: alecdavis 2. in chan_sip.c: /* just confirm that we recieved the packet. */ Have you applied any patches? Another thing to test: where did the string come from at run-time? grep vm-recieved /usr/sbin/asterisk /usr/lib/asterisk/modules/*.so -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SkypeForAsterisk
srinivas Antarvedi wrote: Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. I don't see the need for Asterisk in this scenario. You can do this within the Skype network with permanent call forwarding. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Which of the 3 Skype Users is the s4a account? Current issue with s4a is that you never see who is actually calling. You always see only the skype user on the s4a server. Like in the old days without CLIP/CLIR. Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
2009/12/9 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Dec 09, 2009 at 07:57:52AM +0100, Olivier wrote: What's the output of: lspci -v -nn -s 08:00.0 # lspci -v -nn -s 08:00.0 08:00.0 ISDN controller [0204]: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] [1397:16b8] (rev 01) Subsystem: Cologne Chip Designs GmbH Device [1397:b552] Flags: medium devsel, IRQ 10 I/O ports at ec00 [size=8] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel modules: hfcmulti, wcb4xxp Hmm... you're right. This seems to be a typo in PCI.pm. Fixed, thanks. Great !! Thanks a lot !! I would be very happy to help to check this fix, if you find this of any use ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording from billsec
Daniel Stefanus wrote: I want to rebuild my mixmonitor file.But this time I just want the recording is from the time when the client answer the call,not from the beginning. Anybody can help? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Use option b b - Only save audio to the file while the channel is bridged. *does not include conferences* -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure DAHDI with TDM410 for analog
Hello, I am now working on a project which need data modem calls passing trough asterisk with a TDM card. But i found out (as Mr Greg woods) that it was impossible. So know i am trying doing this using an E1/T1 card (TE110P). I am still working on it so we can help each other to find a solution to this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_voicemail. Help me to find typo source ...
2009/12/9 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Dec 09, 2009 at 08:54:13AM +0100, Olivier wrote: Hi, In /var/log/asterisk/full (Asterisk 1.6.2-rc6), I can see : [Dec 8 15:02:17] VERBOSE[10283] config.c: == Parsing '/var/spool/asterisk/voicemail/default/103/INBOX/msg.txt': [Dec 8 15:02:17] VERBOSE[10283] config.c: == Found [Dec 8 15:02:17] VERBOSE[10283] file.c: -- SIP/103-0784 Playing Call mydialer:030784 'vm-message.gsm' (language 'fr') [Dec 8 15:02:18] WARNING[10283] file.c: File vm-recieved does not exist in any format [Dec 8 15:02:18] WARNING[10283] file.c: Unable to open vm-recieved (format 0x8 (alaw)): No such file or directory [Dec 8 15:02:18] WARNING[10283] say.c: Unable to play message vm-recieved Strangely, I can't find any line in *.c files containing such vm-recieved string. If I add ln -s vm-received vm-recieved in appropriate directory, those error messages disappear. Though I've found a work around, I would be very happy to pin point root cause but I'm a bit lost in the amount of files in Asterisk code. Interesting. Grepping for 'recieve' in 1.6.2.0-rc6 gives only two hits: 1. in the Changelog: 009-07-27 20:33 + [r209234] David Brooks dbro...@digium.com * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces recieved with received and initilize with initialize (closes issue #15571) Reported by: alecdavis I'm aware of this patch which is (should be) included in 1.6.2-rc2. At the time, I also met this bug (I had to add a symbolic link named vm-recieved to work arount it) but as I was unable to pinpoint root cause (which module was referring to such vm-recieved file), I didn't get anywhere. 2. in chan_sip.c: /* just confirm that we recieved the packet. */ Have you applied any patches? No, I don't think so ... but I installed asterisk-addons afterwards ... Before that, I also installed spandsp, mISDN, Dahdi-Tools and Dahdi-Linux Another thing to test: where did the string come from at run-time? grep vm-recieved /usr/sbin/asterisk /usr/lib/asterisk/modules/*.so The reply to this command is empty ! grep vm-received/usr/lib/asterisk/modules/*.so replies app_voicemail.so. The machine from which come the earlier mentioned logs, is in production now and I can't easily change its settings. Fortunately, I've got its installation script and I'm planning to play it on another machine to see if I can reproduce this message ... -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi restart kills server
I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2. I restarted for no good reason (I was playing around), but it did worry me that if Dahdi crashed while Asterisk was running that not only Dahdi and Asterisk would crash, but the whole machine too. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, December 09, 2009 1:47 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dahdi restart kills server On Tue, Dec 08, 2009 at 07:06:52PM -0500, Mike wrote: I`ve just experience a dead server, because I ran /etc/init.d/dahdi restart . I had to reboot the server. What version of DAHDI (tools, linux)? What DAHDI hardware (if any) do you have? What do you have on /etc/dahdi/modules ? Should I worry about something not being right in my install, or is there a known problem with doing this while Asterisk is running? The module dahdi cannot be unloaded when Asterisk is running. Thus this restart is pointless. It should not crash the system, IIRC. Why did you restart dahdi? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] automon = *1 one touch recording
On 12/08/09 20:43, Christian Victor wrote: 2009/12/8 Joseph syscon...@gmail.com: After pressing *1 console is not showing anything indicating that the call is being recorded: -- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0, transfer) in new stack ? ? -- SIP/479-1270-680060b0 Playing 'transfer' (language 'en') ? ? -- Executing [...@office-closed:2] Dial(SIP/479-1270-680060b0, SIP/11IAX2/iaxy-322|30|rwW) in new stack ? ? -- Called 11 ? ? -- Called iaxy-322 ? ? -- Call accepted by 10.0.0.108 (format ulaw) ? ? -- Format for call is ulaw ? ? -- SIP/11-007855f0 is ringing ? ? -- IAX2/iaxy-322-6005 is ringing ? ? -- SIP/11-007855f0 answered SIP/479-1270-680060b0 ? ? -- Hungup 'IAX2/iaxy-322-6005' ? == Spawn extension (office-closed, 11, 2) exited non-zero on 'SIP/479-1270-680060b0' Did you make sure that your telephone actually sends the DTMF tones (the right way)? It seems that asterisk does not recognise incoming DTM or your verbosity level is not high enough. Chris Yes, I can hear the tones as I'm pressing the *1, it has to do with the timing, pressing the buttons with two finger is quicker than with one. Can someone provide where is the timing Francesco in previous post mentioned. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting problem with IP's
Just a guess, but the connection probably went from full to half duplex. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, December 08, 2009 8:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Interesting problem with IP's Have a trunk 1.4 asterisk, running on centos on the lan at work. A long story, but we had the entire work network on a public address range (90.1.0.x), going to a firewall, then out to the net. At home (192.168.1.x network) I have a router that connects to the firewall via a vpn tunnel. All was great. My cisco 7960 (192.168.1.100) was able to register with the asterisk server on 90.1.0.76 - and there was no audio problems whatsoever. I also must stress that I had nat=no and no nat-specific flags set in asterisk. However,the day came where the techs decided that we should be on a private internal network, and moved all of the devices onto a 10.0.x.x internal network. Needless to say, it wasn't an easy task. Now, although my vpn is connected to the new network, and I can access all of the machine as I used to be able to, I now only have 1-way audio on my phone !! (I can hear, and it gets progressively worse,the other party cannot hear me) Why would this have changed ? Do I need to do nat stuff now ? and why ? Interesting. Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure DAHDI with TDM410 for analog
On Wed, 2009-12-09 at 05:34 -0600, mos...@infolog.mr wrote: I am now working on a project which need data modem calls passing trough asterisk with a TDM card. But i found out (as Mr Greg woods) that it was impossible. Just to be clear about this: don't use me as an authoritative reference for this. I am just repeating what I was told when I asked the same question a while back. --Greg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting problem with IP's
snip Just a guess, but the connection probably went from full to half duplex. /snip Full vs. Half duplex networking would NOT cause half duplex phone calls. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Asterisk and SPA-3000
Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at that time actually works but disconnects the original extension (and transfers the PSTN leg to the new extension as normally). At the CLI there is nothing but a new incoming call from the SPA, exactly as the original call. It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does anyone know what could be causing this problem? -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting problem with IP's
This probably isn't a good guess either, but could OP have just 5060 open and not the other ranges * needs to run the call? Since he has one-way connection, I'm guessing not... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Wednesday, December 09, 2009 8:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Interesting problem with IP's snip Just a guess, but the connection probably went from full to half duplex. /snip Full vs. Half duplex networking would NOT cause half duplex phone calls. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting problem with IP's
HaHa!. That is so funny, made me splurt my coffee over the keyboard. lol Julian 2009/12/9 Danny Nicholas da...@debsinc.com: Just a guess, but the connection probably went from full to half duplex. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Tuesday, December 08, 2009 8:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Interesting problem with IP's Have a trunk 1.4 asterisk, running on centos on the lan at work. A long story, but we had the entire work network on a public address range (90.1.0.x), going to a firewall, then out to the net. At home (192.168.1.x network) I have a router that connects to the firewall via a vpn tunnel. All was great. My cisco 7960 (192.168.1.100) was able to register with the asterisk server on 90.1.0.76 - and there was no audio problems whatsoever. I also must stress that I had nat=no and no nat-specific flags set in asterisk. However,the day came where the techs decided that we should be on a private internal network, and moved all of the devices onto a 10.0.x.x internal network. Needless to say, it wasn't an easy task. Now, although my vpn is connected to the new network, and I can access all of the machine as I used to be able to, I now only have 1-way audio on my phone !! (I can hear, and it gets progressively worse,the other party cannot hear me) Why would this have changed ? Do I need to do nat stuff now ? and why ? Interesting. Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual Phone for CDR Logging
Hi,
I found a solution myself I want to share it.
The pjsua SIP client from http://www.pjsip.org does the trick.
You can run it parallel to the asterisk server, ring for a while and hang up
then.
You just need to change one line in the source code to run it in background:
In pjsip-apps/src/pjsua/pjsua_app.c replace console_app_main(uri_arg); with
while (1) {sleep(500);}
Now ./configure and make dep make clean make und run the sip client
from pjsip-apps/bin
./pjsua-x86_64-unknown-linux-gnu --config-file=ringme.local.cfg
Here is my ringme.local.cfg:
--id sip:you sip account@you ip
--registrar sip:you ip
--realm *
--username you sip account
--password your password
--local-port=5061
--null-audio
--log-file=log.txt
--max-calls=32
--auto-answer=180
--duration=10
You need local-port to run beside asterisk.
auto-answer=180 sends a ringing responds.
Greetings
Philipp
Philipp Roos [Inlogia GmbH] schrieb:
Hi,
I am new to the list, so I hope my questions aren't too stupid.
I am using Asterisk 1.4.21.2 and already set it up to use realtime, so a CDR
for an incoming SIP call is written in my mysql database. This works fine.
The problem is that I don't want to have my phone ringing all the time. I
just need a CDR of everyone how is calling me and to read out the CDR from my
PHP script. I tried to replace the Dial(SIP/6000|30) command in the
extensions table by Ringing(),Wait(5),Busy() but now no CDR entry is created.
Same with Ringing(),Wait(5),Hangup(). Looks like I need a Dial() command for
CDR.
How can I create a virtual phone of some kind, so I get a CDR entry without
actually accepting the call.
Thanks in advance!
Greetings Philipp
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[asterisk-users] AEX800P on HP Prolaint ML115 - Error on Module Load -
Hi list: I'm having problems, with a AEX800P card when plugin on HP ML115 G5 Server, when i load the mdule (wctdm24xxp), it loads with error con dmesg, that say that is a kernel bug of invalid OPCODE 000 or something like that. If i plug the card on SUN Server, i don't have problems with the card, but the server's nvidia video card stop working :-S Any idea? Thanks. -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 alvaro.par...@xmarts.com.mx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A101DE with Dell PE 2850
Thanks a lot for that info Christian. Date: Tue, 8 Dec 2009 20:14:02 +0100 From: Christian Victor christ...@victormedia.de Subject: Re: [asterisk-users] Sangoma A101DE with Dell PE 2850 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 9b9941b90912081114w3db968f9ke2b4ce2d15622...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 2009/12/8 Ricardo Melendez rmelen...@utep.com.mx: First I see at sangoma page that A101DE is PCI-Express? (I think ?x1 for the size of the connector) Yes, it is PCIe x1. There is an A101D wich is PCI(-X). for PCI Express one x4 lane width one x8 lane width I can connect the card to any of the slots?, or only to PCI-Express Slots? (is compatible the card with x4 and x8 PCI-Express slots?) Yes, the A101DE runs in PCIe x4 or x8 and the A101D will run in PCI or PCI-X Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi and Junghanns QuadBRI
Tzafrir, Would you say latest trunk (revision 7672) includes a fix for those OctoBRI boards ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi restart kills server
On 12/09/2009 05:39 AM, Mike wrote: I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2. I restarted for no good reason (I was playing around), but it did worry me that if Dahdi crashed while Asterisk was running that not only Dahdi and Asterisk would crash, but the whole machine too. Mike, Could you open an issue on the issue.asterisk.org, along with any information from your system logs around the time of the crash? Restarting DAHDI should definitely not crash your system and I would be interested in seeing any additional information you have from the crash. Thanks, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help/suggestions for DialPlan
I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime to directly dial that extension. I am setting up extensions to be 3 digits long, all starting with 6 (ie. 601, 602, 603, etc.) My current setup requires a single digit to be pressed before going to that context in the Dialplan. Its all working fine with the exception of direct employee extension dialing. What I have done is to create a context for '6'. In other words, I wait for a single keypress at the main menu and if I get a 6, I then transfer to get the next 2 digits which represents the employee ID and then transfer to that person. But what is happening is that the speed in which Asterisk is picking up the digits from the caller isn't fast enough for some reason, and is often missing the first digit (6) and then processing on some other digit that it picks up on. I don't want to have to change our recorded audio menu since that was done professionally and was expensive, but just want it so that the incoming keys pressed are processed instantly. I'm using a TDM400 from an analog line for the incoming menu, and then transferring to our internal LAN once the extension is chosen via SIP. This is the context of my 'main menu': ; Calls during business hours exten = s,1,Set(TIMEOUT(digit)=1) exten = s,n,Wait(1) exten = s,n,Background(1-MainMenu) exten = s,n,WaitExten(3) exten = s,n,Goto(ts_operator,s,1) I have the context for extensions handled like this: ; User pressed 6 - start of extensions [ts_extensions] exten = s,1,Set(TIMEOUT(digit)=2) exten = s,n,WaitExten(10) exten = s,n,Hangup() exten = t,1,Hangup() exten = i,1,Goto(ts_start,s,1) exten = 01,1,VoiceMail(6...@edgeneering) exten = 01,n,Hangup() exten = 02,1,VoiceMail(6...@edgeneering) exten = 02,n,Hangup() exten = 03,1,VoiceMail(6...@edgeneering) exten = 03,n,Hangup() (I'm dummying out the transfer to SIP phone parts with VoiceMails here for demo purposes) What am I doing wrong that is stopping correct identification of the digits entered by the caller? Thanks in advance for any assistance. Myles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePBX IP Phone Recommendations
Looking for IP Phone recommendation to be used with FreePBX. Approx. 50 IP Phones, the FreePBX behind router. So good NAT support needed. FreePBX is installed from the iso image available in AsteriskNOW website. Sorry for the cross post, hoping that the 2 lists provide 2 different perspective. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help/suggestions for DialPlan
If you add option 6 to the menu for the first position and use the
read command for the 2 last position and use a second line that looks
something like:
exten 6,n,Dial(SIP,6${ENTERED_NUMBER},20,t)
it should work.
The {ENTERED_NUMBER} should be the variable filled with the read
command.
Don't forget to add some lines to catch wrong numbers and time outs
Hope this helps :-)
Erik
On 9 dec 2009, at 18:53, Myles Wakeham wrote:
I am revising our DialPlan strategy for our Asterisk system (1.4.2)
and
looking for some info on 'best practices' for this. Here's what I'm
trying to do:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that
extension.
I am setting up extensions to be 3 digits long, all starting with 6
(ie.
601, 602, 603, etc.)
My current setup requires a single digit to be pressed before going to
that context in the Dialplan. Its all working fine with the exception
of direct employee extension dialing.
What I have done is to create a context for '6'. In other words, I
wait
for a single keypress at the main menu and if I get a 6, I then
transfer
to get the next 2 digits which represents the employee ID and then
transfer to that person.
But what is happening is that the speed in which Asterisk is picking
up
the digits from the caller isn't fast enough for some reason, and is
often missing the first digit (6) and then processing on some other
digit that it picks up on. I don't want to have to change our
recorded
audio menu since that was done professionally and was expensive, but
just want it so that the incoming keys pressed are processed
instantly.
I'm using a TDM400 from an analog line for the incoming menu, and then
transferring to our internal LAN once the extension is chosen via SIP.
This is the context of my 'main menu':
; Calls during business hours
exten = s,1,Set(TIMEOUT(digit)=1)
exten = s,n,Wait(1)
exten = s,n,Background(1-MainMenu)
exten = s,n,WaitExten(3)
exten = s,n,Goto(ts_operator,s,1)
I have the context for extensions handled like this:
; User pressed 6 - start of extensions
[ts_extensions]
exten = s,1,Set(TIMEOUT(digit)=2)
exten = s,n,WaitExten(10)
exten = s,n,Hangup()
exten = t,1,Hangup()
exten = i,1,Goto(ts_start,s,1)
exten = 01,1,VoiceMail(6...@edgeneering)
exten = 01,n,Hangup()
exten = 02,1,VoiceMail(6...@edgeneering)
exten = 02,n,Hangup()
exten = 03,1,VoiceMail(6...@edgeneering)
exten = 03,n,Hangup()
(I'm dummying out the transfer to SIP phone parts with VoiceMails here
for demo purposes)
What am I doing wrong that is stopping correct identification of the
digits entered by the caller?
Thanks in advance for any assistance.
Myles
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Re: [asterisk-users] Need help/suggestions for DialPlan
Have you tried...
exten = 6XX,1,Dial(SIP,${EXTEN},20,t)
--
Jarrod Lash, jar...@fed-com.com
Federated Communications, LLC.
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Wed, Dec 9, 2009 at 12:53 PM, Myles Wakeham my...@techsol.org wrote:
I am revising our DialPlan strategy for our Asterisk system (1.4.2) and
looking for some info on 'best practices' for this. Here's what I'm
trying to do:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
I am setting up extensions to be 3 digits long, all starting with 6 (ie.
601, 602, 603, etc.)
My current setup requires a single digit to be pressed before going to
that context in the Dialplan. Its all working fine with the exception
of direct employee extension dialing.
What I have done is to create a context for '6'. In other words, I wait
for a single keypress at the main menu and if I get a 6, I then transfer
to get the next 2 digits which represents the employee ID and then
transfer to that person.
But what is happening is that the speed in which Asterisk is picking up
the digits from the caller isn't fast enough for some reason, and is
often missing the first digit (6) and then processing on some other
digit that it picks up on. I don't want to have to change our recorded
audio menu since that was done professionally and was expensive, but
just want it so that the incoming keys pressed are processed instantly.
I'm using a TDM400 from an analog line for the incoming menu, and then
transferring to our internal LAN once the extension is chosen via SIP.
This is the context of my 'main menu':
; Calls during business hours
exten = s,1,Set(TIMEOUT(digit)=1)
exten = s,n,Wait(1)
exten = s,n,Background(1-MainMenu)
exten = s,n,WaitExten(3)
exten = s,n,Goto(ts_operator,s,1)
I have the context for extensions handled like this:
; User pressed 6 - start of extensions
[ts_extensions]
exten = s,1,Set(TIMEOUT(digit)=2)
exten = s,n,WaitExten(10)
exten = s,n,Hangup()
exten = t,1,Hangup()
exten = i,1,Goto(ts_start,s,1)
exten = 01,1,VoiceMail(6...@edgeneering)
exten = 01,n,Hangup()
exten = 02,1,VoiceMail(6...@edgeneering)
exten = 02,n,Hangup()
exten = 03,1,VoiceMail(6...@edgeneering)
exten = 03,n,Hangup()
(I'm dummying out the transfer to SIP phone parts with VoiceMails here
for demo purposes)
What am I doing wrong that is stopping correct identification of the
digits entered by the caller?
Thanks in advance for any assistance.
Myles
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Re: [asterisk-users] Dahdi and Junghanns QuadBRI
On Wed, Dec 09, 2009 at 05:59:36PM +0100, Olivier wrote: Tzafrir, Would you say latest trunk (revision 7672) includes a fix for those OctoBRI boards ? The only change was in dahdi-tools . As could be clearly seen from the output of lspci, wcb4xxp already handles this device. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
I am setting up extensions to be 3 digits long, all starting with 6 (ie.
601, 602, 603, etc.)
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,goto(support,s,1)
exten = 3,3,goto(customer-service,s,1)
exten = 8,3,goto(directory,s,1)
exten = _x,3, hangup()
exten = _6xx,3, goto(dial-by-extension,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] automon = *1 one touch recording
Yes, I can hear the tones as I'm pressing the *1, it has to do with the timing, pressing the buttons with two finger is quicker than with one. Can someone provide where is the timing Francesco in previous post mentioned. It's featuredigittimeout (in milli-seconds). Try featuredigittimeout = 1000 That will give you one second, as opposed to the default half second. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Asterisk and SPA-3000
The SPA-3000 is notorious for falsely detecting DTMF tones in regular voice, and when it thinks it hears DTMF, it will produce a short real DTMF tone that's only audible to the SIP side of the device, not the PSTN side, or out of band SIP DTMF message (dependent on how you have the device setup). I'm guessing you have your IVR setup so that a single keypress triggers it, and the SPA-3000 thinks it hears that key. I would suggest to not have any single key start your IVR, as the SPA-3000 only ever seems to incorrectly detect and generate one DTMF tone at a time (it never generates a sequence, just one random key's worth of DTMF). Hope this helps, Andrew On Wed, Dec 9, 2009 at 7:35 AM, Ivan Stepaniuk i...@albafotonica.com wrote: Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at that time actually works but disconnects the original extension (and transfers the PSTN leg to the new extension as normally). At the CLI there is nothing but a new incoming call from the SPA, exactly as the original call. It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does anyone know what could be causing this problem? -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't restart asterisk from script
I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] automon = *1 one touch recording
On 12/09/09 13:40, Kai-Uwe Jensen wrote: Yes, I can hear the tones as I'm pressing the *1, it has to do with the timing, pressing the buttons with two finger is quicker than with one. Can someone provide where is the timing Francesco in previous post mentioned. It's featuredigittimeout (in milli-seconds). Try featuredigittimeout = 1000 That will give you one second, as opposed to the default half second. Yes, that was it; thank you. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks Maybe you need to escape your quotes (\restart gracefully\) in your script? Just a thought... -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help/suggestions for DialPlan
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
I am setting up extensions to be 3 digits long, all starting with 6 (ie.
601, 602, 603, etc.)
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,goto(support,s,1)
exten = 3,3,goto(customer-service,s,1)
exten = 8,3,goto(directory,s,1)
exten = _x,3, hangup()
exten = _6xx,3, goto(dial-by-extension,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Need help/suggestions for DialPlan
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
I am setting up extensions to be 3 digits long, all starting with 6 (ie.
601, 602, 603, etc.)
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,goto(support,s,1)
exten = 3,3,goto(customer-service,s,1)
exten = 8,3,goto(directory,s,1)
exten = _x,3, hangup()
exten = _6xx,3, goto(dial-by-extension,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Need help/suggestions for DialPlan
Watch out for 'Dial by name' missing first digit, there's a bug in
app_directory, at least with 1.6.1 branches and newer.
Not sure of what release you're testing on.
See:
https://issues.asterisk.org/view.php?id=16409
First DTMF digit is missed if pressed during using your touch tone
keypad... announcement
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, 10 December 2009 9:30 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need help/suggestions for DialPlan
On Wed, 9 Dec 2009, Myles Wakeham wrote:
I have an ACD menu that gives the caller the options as follows:
- Press 1 for sales
- Press 2 for support
- Press 3 for customer service
- Press 8 for a 'Dial by Name' list
or enter the extension number at anytime to directly dial that extension.
I am setting up extensions to be 3 digits long, all starting with 6 (ie.
601, 602, 603, etc.)
Patterns and wildcards are your friend.
Maybe something like:
[example]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,2, answer()
exten = 1,3,goto(sales,s,1)
exten = 2,3,goto(support,s,1)
exten = 3,3,goto(customer-service,s,1)
exten = 8,3,goto(directory,s,1)
exten = _x,3, hangup()
exten = _6xx,3, goto(dial-by-extension,s,1)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Problem with Asterisk and SPA-3000
Andrew Hakman wrote:
The SPA-3000 is notorious for falsely detecting DTMF tones in regular
voice, and when it thinks it hears DTMF...
Thanks Andrew. This is true, we do have false DTMF detections/playback
(Lowering RX gain really helps on this). However this does not seem to
be related. The only way to get to the IVR is through the 's' extension
on the context in which the SPA is registered. The SPA has a built-in
dialplan that is set to something like (S0:s). Asterisk context
shows is the following:
[sip_pstn_linksys]
exten = s,1,NoOp(Caller-ID-number: ${CALLERID(number)})
exten = s,n,NoOp(Caller-ID-name: ${CALLERID(number)})
exten = s,n,Set(CALLERID(name)=Externa SPA PSTN)
exten = s,n,GotoIf($[${CALLERID(num)} = pstn1]?private,1)
exten = s,n,Goto(menu,s,1)
exten = private,1,Set(CALLERID(number)=Privado)
exten = private,n,Goto(menu,s,1)
--
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com
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Re: [asterisk-users] Interesting problem with IP's
Don't forget that many routers treat the designated private address space differently because it assumes the device is being implemented as a border router. In this configuration they block most traffic unless you specifically set rules to permit traffic to flow. -dbc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as a PSTN simulator
Hey folks, I'm from Brazil and I have the following doubt. May I use an asterisk box with some cards to act as an PSTN simutator between a little amount of sites? I will start to think about it in a voice lab for my studies. Kind regards, Vitor -- Vitor Afonso Strabello MSN: vstrabe...@hotmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a PSTN simulator
Depends on how authentic you want your simulation to be. On 12/09/2009 07:49 PM, Vitor Afonso Strabello wrote: Hey folks, I'm from Brazil and I have the following doubt. May I use an asterisk box with some cards to act as an PSTN simutator between a little amount of sites? I will start to think about it in a voice lab for my studies. Kind regards, Vitor -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] show queue's name and other info in incoming call to queue member
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example client's_number - Sales. This problem appears when one member
can belong to couple queues. Work around would be setting calling name with
such information.
If Your phone supports text CLID:
Set(CALLERID(name)=${CALLERID(num) - Sales);
Queue(sales);
If not, You can just add some digit in front/end of CALLERID(num).
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] show queue's name and other info in incoming call to queue member
On Thu, Dec 10, 2009 at 2:54 AM, Atis Lezdins a...@iq-labs.net wrote:
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example client's_number - Sales. This problem appears when one member
can belong to couple queues. Work around would be setting calling name with
such information.
If Your phone supports text CLID:
Set(CALLERID(name)=${CALLERID(num) - Sales);
Ooops, syntax validation was off:
Set(CALLERID(name)=${CALLERID(num)} - Sales);
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Can't restart asterisk from script
Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
But my command works from a bash command line...so something is fishy! I want to solve this mystery -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, December 09, 2009 8:27 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
Interesting...I'll try that. Thanks _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
But the error message in my log shows the error to be from asterisk, so I'm guessing I'm sending a parameter incorrectly to asterisk - which fits with the no quote theory -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy Sent: Wednesday, December 09, 2009 9:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-users@lists.digium.com Hello!
Hi! I saw your profile and would like to get to know you better. Im looking for open, adventurous people, in my area, but we can start here. Email me back at maris...@email-chatting.com . Muah! Marishka ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Database Tables
Hello, We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would like to use the sip,extensions and voicemail in realtime mode. Where can we find the database tables structure for these versions? Thanks, Andreas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
You should replace the single quote with double quote. --Original Message-- From: Michelle Dupuis Sender: asterisk-users-boun...@lists.digium.com To: 'Asterisk Users List' ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't restart asterisk from script Sent: Dec 9, 2009 10:59 PM But the error message in my log shows the error to be from asterisk, so I'm guessing I'm sending a parameter incorrectly to asterisk - which fits with the no quote theory -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy Sent: Wednesday, December 09, 2009 9:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Database Tables
On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote: Hello, We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would like to use the sip,extensions and voicemail in realtime mode. Where can we find the database tables structure for these versions? Thanks, Andreas This is the first place to go: http://www.voip-info.org/wiki/view/Asterisk+RealTime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Database Tables
Thanks Fred, I'm actually there, but I was wondering if the tables there are up to date and if any changes took place. I see all kinds of comments about changes. Original Message Subject: Re: [asterisk-users] Realtime Database Tables From: Fred Posner f...@teamforrest.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, 10 December, 2009 05:26:07 On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote: Hello, We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would like to use the sip,extensions and voicemail in realtime mode. Where can we find the database tables structure for these versions? Thanks, Andreas This is the first place to go: http://www.voip-info.org/wiki/view/Asterisk+RealTime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
I had double quotes originally - and that didn't work -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Wednesday, December 09, 2009 10:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script You should replace the single quote with double quote. --Original Message-- From: Michelle Dupuis Sender: asterisk-users-boun...@lists.digium.com To: 'Asterisk Users List' ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't restart asterisk from script Sent: Dec 9, 2009 10:59 PM But the error message in my log shows the error to be from asterisk, so I'm guessing I'm sending a parameter incorrectly to asterisk - which fits with the no quote theory -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy Sent: Wednesday, December 09, 2009 9:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
Yes, but if asterisk cant find some of its components due to abbreviated path or ... Just run a cron that prints the results from env and compare and see if there is something obvious - there may also be privilege issues BillK On Wed, 2009-12-09 at 22:32 -0500, Michelle Dupuis wrote: I had double quotes originally - and that didn't work -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Wednesday, December 09, 2009 10:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script You should replace the single quote with double quote. --Original Message-- From: Michelle Dupuis Sender: asterisk-users-boun...@lists.digium.com To: 'Asterisk Users List' ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't restart asterisk from script Sent: Dec 9, 2009 10:59 PM But the error message in my log shows the error to be from asterisk, so I'm guessing I'm sending a parameter incorrectly to asterisk - which fits with the no quote theory -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy Sent: Wednesday, December 09, 2009 9:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On Wed, 9 Dec 2009, Michelle Dupuis wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) By cron job do you mean an entry in the-user-executing-asterisk's crontab like one of the following? @daily /usr/sbin/asterisk -r -x restart gracefully @daily /usr/sbin/asterisk -r -x 'restart gracefully' @daily /usr/sbin/asterisk -r -x restart gracefully @daily /scripts/restart-when-convenient.sh The first 2 work, the 3rd doesn't. If you are using something like the 4th, the error is in your script so you should post the script. The difference between the first 2 is that 1 uses double quotes and 2 uses single quotes. Quotes mean pass everything between the quotes as a single argument. Double quotes mean evaluate any expressions or environment variables before passing. Single quotes mean pass everything literally, without evaluation or substitution. Single quotes are better than double quotes because we already know there is nothing to evaluate or substitute in restart gracefully so there is no need to pass the quoted string through the evaluation/substitution code in the shell. The 3rd doesn't work because just restart is passed as the argument to the x option and restart by itself is not a valid Asterisk command. Please paste your crontab if you still need help. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On Wed, Dec 09, 2009 at 03:28:19PM -0600, Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks Maybe you need to escape your quotes (\restart gracefully\) in your script? Huh? As you can see, Asterisk got the string as-intended. No need for funny quoting tricks. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Meetme on XEN virtual machine recording is 2x times faster than normal
hello ALL, i have one virtual machine creating via XEN we running conference on that using SIP channel everything is fine except recording a .wav file is 2x times faster that original voice is there any setting regarding that to improve it? regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
2009/12/9 Michelle Dupuis supp...@ocg.ca I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? I've met a situation where a call remained active tough obviously, no one was talking to anyone. This kept gracefully restart from working. So, would a 'restart now' behave differently ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On Thu, Dec 10, 2009 at 08:02:22AM +0100, Olivier wrote: 2009/12/9 Michelle Dupuis supp...@ocg.ca I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? I've met a situation where a call remained active tough obviously, no one was talking to anyone. This kept gracefully restart from working. So, would a 'restart now' behave differently ? Why would one want a daily sabotage of the system in the first place? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On 12/10/2009 02:07 AM, Tzafrir Cohen wrote: Why would one want a daily sabotage of the system in the first place? An apt question. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
2009/12/10 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Dec 10, 2009 at 08:02:22AM +0100, Olivier wrote: 2009/12/9 Michelle Dupuis supp...@ocg.ca I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? I've met a situation where a call remained active tough obviously, no one was talking to anyone. This kept gracefully restart from working. So, would a 'restart now' behave differently ? Why would one want a daily sabotage of the system in the first place? Reading this thread, I wondered why a gracefully restart wouldn't work in the first place and focused my modest contribution on this. You're right that a daily restart now in not recommended. IMHO, for daily maintenance, taking this specific case (a ghost call is keeping restart gracefully from running) into account is necessary. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] switchvox 305 Appliance
I am new to the list and wanted to get the professionals here input on Switchvox 305 Appliance ? List price is 4k, ouch! Is there a better cost-effective way ? Also feedback (neg/pos) about this appliance. -mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
