[asterisk-users] Newbie Polycom: Cannot Disable Services button
I was able to disable the DND button (no 9) on IP60x by putting the following line in sip.cfg. keys key.scrolling.timeout=1 key.IP_600.9.function.prim=Null/ However, I could not do so for the Services Button (no 29) on IP600 (or Applications button on IP01) keys key.scrolling.timeout=1 key.IP_600.29.function.prim=Null/ It seems like SIP admin guide did not include Services as the function that can be modified. Anyone having the same experience? ___ -- 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] module reload CLI Asterisk question
Hello Alejandro, I'm not sure if it related, but I saw this behavior when the Asterisk service was started without using the init script. Regards Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Monday, May 12, 2008 1:30 PM To: Asterisk Users Mailing List Subject: [asterisk-users] module reload CLI Asterisk question Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk files very well. Always I enter the CLI (with asterisk -r) and when I make a change after that I execute module reload and everything is OK. But a few days ago, without make any change, I execute module reload from within CLI and the terminal turn into black color and the color of the letters was white (exactly the opposite to the normal colors). I think because I get some warning and notice message like these: [[May 12 10:19:10] NOTICE[6265]: cdr.c:1362 do_reload: CDR simple logging enabled. [May 12 10:19:10] NOTICE[6265]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' [May 12 10:19:10] WARNING[6265]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4090 pbx_load_module: Starting AEL load process. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4097 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder) [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4105 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4108 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4110 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4113 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4116 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. After that I test the system and it work OK. What can be the problem ??? Is it a normal situation ??? Thanks a lot. Alejandro ___ -- 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] Is there a way to have Manager Bridge Channel without being connected
On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, Is there a way to have Manager Bridge Channel to the specified extension without the channel being connected. In the current scenario the channel only bridges once the call get connected, it does not bridge when any service provider (telco) message is played. I want to record all call originated by manager even if a telco message is played. I think the only way you'll be able to do that is by capturing the RTP packets with a wireshark or tcpdump. The message from your telco sounds like it's early media (in the SIP World it's called Call Progress media) and as you point out it's generated without answering the call. Most of the time the early media is going to be a ring tone and hence very uninsteresting to record so I suspect no one has bothered to write an application for it. Regards, Greyman. ___ -- 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 stops MOH on transfer
Hello, i´ve a problem i dont find the reason for. An incoming call coming over iax is connected to a Sip phone. Until the phone picks up the call i could hear moh without problems. Then the phone sets the call on hold and opens another call to another extension. The incoming call hears the Hold music and also the call to the other extension hears another moh. Everything works so far as it should. Now the two calls are transfered so the incoming call is switched to the other extension. At this point the Moh on both channels stops which should be. The incoming call should hear the moh, or a ringing, but just silence. i do the dial command with the m option. i could paste logs if necessary. The version which runs here is a Asterisk 1.2.26.2 and a realtime configuration. best regards Steve Smith -- Für weitere Fragen stehen wir gerne unter [EMAIL PROTECTED] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // [EMAIL PROTECTED] // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- 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 stops MOH on transfer
2008/5/13 Stefan Schmidt [EMAIL PROTECTED]: Hello, i´ve a problem i dont find the reason for. An incoming call coming over iax is connected to a Sip phone. Until the phone picks up the call i could hear moh without problems. Then the phone sets the call on hold and opens another call to another extension. The incoming call hears the Hold music and also the call to the other extension hears another moh. Everything works so far as it should. Now the two calls are transfered so the incoming call is switched to the other extension. At this point the Moh on both channels stops which should be. The incoming call should hear the moh, or a ringing, but just silence. i do the dial command with the m option. i could paste logs if necessary. The version which runs here is a Asterisk 1.2.26.2 and a realtime configuration. I found the same issue, and a similar issue with transferring a call received out of a queue. Both issues exist in 1.2, and a friend of mine kindly re-checked this and found that it was fixed/changed in 1.4 Regards, Steve ___ -- 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] 3U server chassis Digium TE405P?
On Mon, 12 May 2008, Steve Totaro wrote: You can put a TE405 in a 1 server (horizontally of course). I've just built a bit of an experimental system with 2 PCI cards in a 1U box. Quite a neat little system. The cards are a TDM400 and a TE120P. Will they work? Well, they seem to, so-far... Be intersting to see how it behaves under load and I'll get a chance to find out in the next few days. Gordon ___ -- 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] exten = pattern match query
2008/5/12 Steve Davies [EMAIL PROTECTED]:
Hi,
I read the WiKi, which implied there was a way of working around this,
but the HTML nature of the WiKi seems to have destroyed some of the
output so I cannot see the correct answer...
I would like to match a special case of a number dialled 0x, now
normally I would simply do:
exten = _0x.,1,NoOp(Got Hex dialling)
But the X pattern match is case-insensitive, so the above pattern
will match any 3 or more digit number starting with a zero. I suspect
that the answer may be:
exten = _0[x].,1,NoOp(Got Hex dialling)
Could someone confirm this for me?
For the archives, in 1.2.28, the following is true...
_0x.
matches a 0, followed by a digit ('x' is NOT a digit) followed by anything
_0[x].
matches a 0 followed by a lower case X followed by anything.
I should have just tried it for myself in the first place :)
Steve
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Re: [asterisk-users] cannot get calls with Tellfree brazilian provider
If you want receive calls with the user B, must change the type to USER. Please make tests and let we know the results. Regards On Tue, May 13, 2008 at 7:29 AM, gincantalupo [EMAIL PROTECTED] wrote: Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the sip show registry command says user B is registered. In my sip.conf I have: register = userB:[EMAIL PROTECTED]/userB [userB] type = peer nat = yes insecure = very canreinvite = yes qualify = yes ; Authentication and channel options secret = xx username = yyy authuser = yyy authname = yyy fromuser = yyy host = sip.tellfree.net fromdomain = sip.tellfree.net port = 5060 context = tellfreein Is there anybody (Brazilian or not) who can give me a hint, please? Thank you. Giorgio ___ -- 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 -- Gabriel Lopes ___ -- 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] Extension Auto Fall through help when matching fails.
Hi,
I'm having a little difficulty with my extensions setup.
What I'm trying to do is to have a PBX where I can call in to check
mail and call-out using the attached mobile or SIP phones.
If someone I know calls then they can be forwarded to me.
if it is someone I don't know then just ring the local sip numbers and
go to voicemail.
So what I did was something like the following. Note I just wrote this
from memory as I don't have remote access to my machines from work. So
apologies if the Goto grammar is not quite accurate.
The problem that I am facing is that when the callerid works (X100P in
UK / BT with patches) then it jumps to the owners and goes in to DISA.
Although I may have echo issues as any button presses don't seem to be
recognised by DISA(That is a different problem).
If the callerid fails or I withhold my number I would expect it to
drop down to s,n,Goto(call-house,1) but I get an Auto fall through
message and it just rings out.
What have I got wrong? I just want an easy way to match two sets of
numbers 'owners' and 'friends' all other callers should hit the last
'call-house' jump.
TIA
Martin
[globals]
house-numbers=SIP/officeSIP/lounge
my-mobile-number=07123456789
[default]
exten = s,1,Verbose(Incoming call ${CALLERID(num)})
exten = s/${my-mobile-number},n, Goto(owners,1)
; if anyone has an easier way than copying this line for all my
friends numbers then that would be great!
exten = s/$(friends-number),n,Goto(call-me,1)
exten = s,n,Goto(call-house,1)
exten = owners,1,DISA(no-password)
exten = owners,n,Hangup
exten = call-me,1,Dial(${house-numbers}Mobile/3230/${my-mobile-number})
exten = call-me,n,Hangup
exten = call-house,1,Dial(${house-numbers})
exten = call-house,n,Hangup
--
Martin Ritchie
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Re: [asterisk-users] 3U server chassis Digium TE405P?
On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 12 May 2008, Steve Totaro wrote: You can put a TE405 in a 1 server (horizontally of course). I've just built a bit of an experimental system with 2 PCI cards in a 1U box. Quite a neat little system. The cards are a TDM400 and a TE120P. Will they work? Well, they seem to, so-far... Be intersting to see how it behaves under load and I'll get a chance to find out in the next few days. Gordon Gordon, It may work. Just a benchmark from my experience, a sing core HP DL 360 @ ~3ghz and two gigs of RAM gave me ~75 CPU usage in top. This box was used simply as a PSTN to SIP (ULAW) gateway with just the required features and programs. I was using a Sangoma four port T1 card with 95 simultaneous calls (NFAS). Thanks, Steve Totaro ___ -- 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] Control of individual call legs
Hello , is it possible to control multiple legs (channels) of a call individually, ie. call 1 -- incoming call connected to IVR call 2 -- outgoing call to party a made via manager interface call 3 -- outgoing call to party b made by call-script I would like to allow the caller on call1 to be able to decide if they want to be connected to call2, call3, or generate an additional call4 for there use, and I don't want to use a meeting room. Thanks for any tidbits! 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
Re: [asterisk-users] Paging for analoge devices
Hi Steve; If we give a look for the link http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page there are some topics not cleared to complete the paging senario: 1) Why to use Set(_ALERT_INFO=RA)? And how can I know what each device take? 2) Does the following work for Polycom: SIPAddHeader(Call-Info:sip:domain\;answer-after=0) 3) Can I do paging without use Page and only by using Dial (but I will set the previous paramters as needed)? 4) Any configuration need to be done on the Meet Me conference to have the Paging working fine? Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: Bilal, Providing your Asterisk box has onboard sound or you can add a card or even USB sound then you will just use your Asterisk server to act as a phone basically. It even has autoanswer so it should be perfect. I think you have enough options now to act. http://www.voip-info.org/wiki-Asterisk+tips+console Then you need to either feed it into an AMP (Bogen or whatever) or buy speakers with built in AMPs and volume control. Thanks, Steve Totaro On Tue, Apr 29, 2008 at 3:43 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Steve; Using the computer sound card is a very nice solution, but how to connect it to the pbx? How to connect it to the fxs port and give it an extension? Do I need an phone? Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: Bilal, Geez, yes, that is how overhead paging works. You can even buy the speakers with built in AMPs and volume control and you could use your computer's sound card. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 6:30 PM, bilal ghayyad [EMAIL PROTECTED] wrote: I will try to see the analoge and how it can be done. About the overhead, do u mean the AMP and speakers or what? Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: Bilal, So you want to page through your analog phones, no overhead paging. I doubt this is possible. I think your options are an IP phones such as the Polycom that supports paging or using an AMP and speakers usually mounted high on the wall or flush with a tile ceiling. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 11:58 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Steve; Already we need to use the same analoge phones that is connected to the fxs ports, to do paging to it. What overhead? Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: Bilal, You will have great luck with Polycom then, not sure about others. Why not go for overhead paging? It will be much easier. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:24 AM, bilal ghayyad [EMAIL PROTECTED] wrote: And auto answer also will be needed for IP Phones? Regards Bilal --- Steve Totaro [EMAIL PROTECTED] wrote: Bilal, Sorry to reply to my reply but if you can register multiple accounts and setup auto answer on one of those accounts, it could work. The problem is, I am not sure if there are any ATAs that have this ability. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:06 AM, Steve Totaro [EMAIL PROTECTED] wrote: Bilal, No, I do not think that you can make this work. You would obviously need auto answer or how else would the audio come out of the speakers? I thought you were talking about overhead paging. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 8:49 AM, === message truncated === Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Queuing if no one available to answer
Hi list; Any one can advise how to put the caller in the queue in case no one available to take his call? All are busy (having calls)? Regards Bilal ___ -- 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] Control of individual call legs
On Tue, May 13, 2008 at 8:35 AM, David Boyd [EMAIL PROTECTED] wrote: Hello , is it possible to control multiple legs (channels) of a call individually, ie. call 1 -- incoming call connected to IVR call 2 -- outgoing call to party a made via manager interface call 3 -- outgoing call to party b made by call-script I would like to allow the caller on call1 to be able to decide if they want to be connected to call2, call3, or generate an additional call4 for there use, and I don't want to use a meeting room. Thanks for any tidbits! Dave Check out app_bridge Thanks, Steve Totaro ___ -- 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] Is there a way to have Manager Bridge Channel without being connected
On Tue, May 13, 2008 at 4:22 AM, Grey Man [EMAIL PROTECTED] wrote: On Mon, May 12, 2008 at 9:44 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, Is there a way to have Manager Bridge Channel to the specified extension without the channel being connected. In the current scenario the channel only bridges once the call get connected, it does not bridge when any service provider (telco) message is played. I want to record all call originated by manager even if a telco message is played. I think the only way you'll be able to do that is by capturing the RTP packets with a wireshark or tcpdump. The message from your telco sounds like it's early media (in the SIP World it's called Call Progress media) and as you point out it's generated without answering the call. Most of the time the early media is going to be a ring tone and hence very uninsteresting to record so I suspect no one has bothered to write an application for it. Regards, Greyman. Simple answer, www.orecx.com Thanks, Steve Totaro ___ -- 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] MeetMeAdmin() working problem
Hello users, Actually i am planning to setup a conference system i have following dialplan [default] exten = 12345,1,MeetMe(1234|X) exten = 12345,2,Hangup() exten = 1,1,MeetMeAdmin(1234|M|user1) exten = 1,2,GoTo(12345|1) exten = 2,1,MeetMeAdmin(1234|m|user1) exten = 2,2,GoTo(12345|1) exten = 3,1,MeetMeAdmin(1234|k|user1) exten = 3,2,GoTo(12345|1) exten = 4,1,MeetMeAdmin(1234|N) exten = 4,2,GoTo(12345|1) exten = 5,1,MeetMeAdmin(1234|n) exten = 5,2,GoTo(12345|1) exten = 6,1,MeetMeAdmin(1234|K) exten = 6,2,GoTo(12345|1) Actually users login into the conference system by dialing in 12345 to enter into conference 1234 and the admin presses 1,2,3,4,5,6 to implement features of conference respectively Mute single user unMute single user Kick single user Mute total conference unMute total conference Kick total conferece while extensions 4,5,6 working fine but individual users mute,unmute,kick(1,2,3 options) not working and the CLI showing specified user not found can anybody helpme out not using any zaptel drivers using only ztdummy Thanks in advance Srinivas Antarvedi ___ -- 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] cannot get calls with Tellfree brazilian provider
Hi Gabriel, it works! I have tried FRIEND in previous tests but probably it did not work because of other mistakes which disappeared when I changed the parameters... Thank you!!! Giorgio Gabriel Lopes wrote: If you want receive calls with the user B, must change the type to USER. Please make tests and let we know the results. Regards On Tue, May 13, 2008 at 7:29 AM, gincantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the sip show registry command says user B is registered. In my sip.conf I have: register = userB:[EMAIL PROTECTED]/userB http://userB:[EMAIL PROTECTED]/userB [userB] type = peer nat = yes insecure = very canreinvite = yes qualify = yes ; Authentication and channel options secret = xx username = yyy authuser = yyy authname = yyy fromuser = yyy host = sip.tellfree.net http://sip.tellfree.net fromdomain = sip.tellfree.net http://sip.tellfree.net port = 5060 context = tellfreein Is there anybody (Brazilian or not) who can give me a hint, please? Thank you. Giorgio ___ -- 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 -- Gabriel Lopes ___ -- 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] 3U server chassis Digium TE405P?
On Tue, 13 May 2008, Steve Totaro wrote: On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 12 May 2008, Steve Totaro wrote: You can put a TE405 in a 1 server (horizontally of course). I've just built a bit of an experimental system with 2 PCI cards in a 1U box. Quite a neat little system. The cards are a TDM400 and a TE120P. Will they work? Well, they seem to, so-far... Be intersting to see how it behaves under load and I'll get a chance to find out in the next few days. Gordon Gordon, It may work. It's just an experiment, but who knows :) Just a benchmark from my experience, a sing core HP DL 360 @ ~3ghz and two gigs of RAM gave me ~75 CPU usage in top. This box was used simply as a PSTN to SIP (ULAW) gateway with just the required features and programs. I was using a Sangoma four port T1 card with 95 simultaneous calls (NFAS). This is for a small office of 30 people - only 10 channels of the PRI are lit. The TDM card is for outgoing calls to a 2-port Premicell unit (analogue GSM adapter with LCR to mobile phones) and for 2 fax machines - which is the only thing I'm a shade concerend about - the analogue FAX through the TDM board and back out via the PRI board.. (incoming faxes are handled by spandsp/RxFAX and sent via email) And this in only a mere 1.3GHz VIA board too. 384MB of free RAM (512 in total, but the OS lives in a 128MB ramdisk) No transcoding and everything is custom compiled/built. (Asterisk 1.2) I actually wanted them to go for a Xorcom channel bank, as they have a door opener/bell-push to manage somehow too, but they weren't keen on it, and the Xorcom is a bit over kill - if only they did a 2+2 or even 4+4 (FXO/FXS) unit rather than an 8+8 ... (or even a stand-alone IO port!) I'll let you know how it fares :) Gordon ___ -- 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] Calls on E1 TDMoE span are dropped at random
On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. Thanks for your help! Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at ___ -- 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] Calls on E1 TDMoE span are dropped at random
Florian Hackenberger wrote: On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. I did not follow the thread, but can this be a timing problem ? It might be that the far end goes into maint mode due to slips, or what ever. Regards Hans ___ -- 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] Queues, monitor-join=yes, and volume
Hi guys, I have few queues configured on my PBX that have: monitor-format=wav monitor-join=yes My problem is that the volume of the clients who are calling into these queues is slightly higher than the volume of the agents. Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? I'm using version 1.2.9, which I think uses soxmix for joining the files? Thanks, Alex ___ -- 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 stops MOH on transfer
Steve Davies schrieb: I found the same issue, and a similar issue with transferring a call received out of a queue. Both issues exist in 1.2, and a friend of mine kindly re-checked this and found that it was fixed/changed in 1.4 Regards, Steve ___ but i think that this only has begun after i have done a upgrade to 1.2.26. Before i had the version 1.2.12 running without this problem. But when it´s resolved in 1.4 then the long way to changing to this version has to be walked sooner than i thought ;) thanks and regards steve -- Für weitere Fragen stehen wir gerne unter [EMAIL PROTECTED] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // [EMAIL PROTECTED] // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- 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] Queues, monitor-join=yes, and volume
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote: Is there any way to modify the volume (either lower the volume of the clients, or increase the volume of the agents) while doing the join of the -in and -out files into one recording? Uh-huh. Read the documentation for soxmix. ___ -- 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] Calls on E1 TDMoE span are dropped at random
On Tue, May 13, 2008 at 9:57 AM, Florian Hackenberger [EMAIL PROTECTED] wrote: On Tuesday 13 May 2008, Steve Totaro wrote: You can be shot several times and not die. I would try resetinterval=never just to be able to to say Not the problem rather than Probably not the problem. I'll do that, although I'm pretty sure that the setting is not the problem as the yellow alarm occured quite often a few minutes after restarting asterisk and the default is 3600 seconds. PRI debug info would be a great help too. The log I sent in the original message contains pri debug messages. I just had another look at it. Thanks for your help! Cheers, Florian -- DI Florian Hackenberger [EMAIL PROTECTED] www.hackenberger.at Ah, didn't even know you could add attachments to postings to the list. It is a little hard to read and quite a bit of info that may or may not be a problem. Does this happen enough, or do you have enough time to sit there and catch the exact log output when it happens? If you can comment out the spans you are not using, that would reduce a bit of output (I assume you have a single E1 and some POTS (although I don't see them configured, but you said in your initial posting that you could dial out on POTS). ERROR[7968]: chan_zap.c:8176 zt_pri_error: !! Got reject for frame 120, retransmitting frame 120 now, updating n_r! and ERROR[7968]: chan_zap.c:8176 zt_pri_error: !! Got I-frame while link state 2 -- Got UA from network peer Link up. looks suspicious. Maybe Red-fone could give you some insight on these errors. If you can narrow it down then I am sure someone can better. Again, comment out spans not in use, set your verbose to 0, and turn on PRI debugging and try to catch only the event/s that correlate with the calls being dropped. I saw Red-fone's products at Astricon, they looked great for failover. Can you describe exactly how you are utilizing it, including LAN/WAN, switches, ping times, and other network central details. TDMoE adds the E (ethernet) component to troubleshooting and I think do to this, it may be very fragile depending on network conditions. Don't make the mistake of just focusing on Asterisk and Zaptel in your troubleshooting process. Thanks, Steve Totaro ___ -- 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 test dialplan w/o a trunk
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
the LAN behind the firewall.
Can I, and if so how do I, set-up sip.conf to force my soft-phone to go
to a specific context when I take it off-hook? (The [Dial/Answer] button
in ZoIPer). Or should I set up an extension that just goes to the context?
I guessing
[613]
...
context=incoming
...
should do it.
I don't have the system on the bench yet but would like to get the
dialplan fairly close the first time. :-)
TIA,
Rod
--
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[asterisk-users] Asterisk-Tag.org conference, May 26th/27th, Berlin, Germany
Asterisk-Tag.org 2008 May 26th/27th Berlin, Germany http://www.asterisk-tag.org http://www.heise.de/open/Marc-Spencer-eroeffnet-den-Asterisk-Tag--/news/meldung/107462 Speakers: * Mark Spencer (Founder of Digium and Inventor of Asterisk) - Digium * Kevin P. Fleming (Director of Software Technologies) - Digium * Michelle Petrone - Digium * Olle E. Johansson (SIP and Asterisk guru) - Edvina * Jay Phillips (Inventor of Adhearsion) - Adhearsion * Randy Resnick aka Randulo (host of the weekly Asterisk talk show) - VoIP Users Conference * Phil Zimmermann (Yes, the PGP guy talks about encrypted VoIP!) - philzimmermann.com * Tim Köhler - snom * Eric Kirchner (Head of Business Development) - Aastra DeTeWe * Stefan Wintermeyer (Geschäftsführer) - Amooma * Philipp Kempgen (Leiter Entwicklung Gemeinschaft) - Amooma * Nenad Corbic (Chief Software Architect, Development Manager) - Sangoma Technologies Corp. * Stephen Bosch (Managing Director) - Vodacomm Voice Data Corporation * Daniel-Constantin Mierla (Co-Founder and Core Developer) - OpenSER * Diana Cionoiu (Founder) - Yate (softswitch) May 26th, German track: * http://www.asterisk-tag.org/wiki/Programm_26.05.2008_(Deutscher_Track) May 26th, English track: * http://www.asterisk-tag.org/wiki/Programm_26.05.2008_(English_Track) May 27th, mixed track: * http://www.asterisk-tag.org/wiki/Programm_27.05.2008_(Mixed_Track) May 27th, Workshops: * http://www.asterisk-tag.org/wiki/Programm_27.05.2008_(Workshops) Tickets include entrance to the Linux-Tag, May 28th-31st http://www.linuxtag.org/2008/ Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- 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] Fwd: [asterisk-dev] Paging intercom extensions
-- Forwarded message -- From: Tilghman Lesher [EMAIL PROTECTED] Date: Tue, May 13, 2008 at 11:20 AM Subject: Re: [asterisk-dev] Paging intercom extensions To: Asterisk Developers Mailing List [EMAIL PROTECTED] On Tuesday 13 May 2008 10:05:19 Gideon Spreeth wrote: The problem comes when we want to make an intercom group page that includes AllPage extensions on more that one server. The page function does not seem like a viable option as you cannot provide it with extension numbers to dial. Does anyone have an idea how to make such a group call possible? Because of the scale of the system and the idea of setting up groups dynamically, it is not feasible to set up a large enough number of extensions. The idea is to call the all page extension with a softphone that can pass the group numbers to it. First of all, this is a -users level question, and you should have posted this question there. And second, you need to look at the Local channel type, as it allows you to do exactly that. -- Tilghman ___ -- 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] How to test dialplan w/o a trunk
Roderick A. Anderson wrote: Can I, and if so how do I, set-up sip.conf to force my soft-phone to go to a specific context when I take it off-hook? This can be done with a analog phone, but I don't believe you can do it on a sip channel. (The [Dial/Answer] button in ZoIPer). Or should I set up an extension that just goes to the context? That's the route I'd follow. 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
[asterisk-users] New Asterisk Deployment - Need some tips
I'll be doing a new Asterisk deployment soon, and would like to gather your thoughts. Here are some items that need to be kept in mind: Support 800 phones (400 of which are analog) Concurrent calls ... ? but need to guess high so that the server can handle this. Voicemail will be required along with sending voice mail attachments to email server. Flash panel for switchboard operator. Needs to be a distributed server design for redundancy and fail-over. Will need to be integrated into an existing PBX until each building is switched over to use the Asterisk servers. If calling 911 from a building among multiple buildings, how can EMS find that person based upon the call? What type of data line should be used in this setup? T1? The physical network will support QOS and the like, so that is not an issue. What type of design/setup do you recommend for this? How about server resources...ie...CPU, RAM, Disk space. How about backups? Does imaging work best if a server were to fail? Any thing else you can think of? _ This email was transferred using an Office free edition of AXIGEN Mail Server. ___ -- 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] DTMF lose with TE-121F
Problem solved turning off echo cancellation. Any known bug? Pepe Aracil escribió: Hello. I'm using asterisk in alarm reception system. The system is DTMF intensive and works well while all concurrent channels are online. But when one channel goes hangup the other channels lose tones while one 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 ___ -- 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] More one way audio...
I am a bit desperate trying to solve this problem. Sorry if I am abusing the list a bit with the same king of question. The problem I am having is very specific which is why it is very difficult to diagnose and fix. Basically an Asterisk server is connected via E1 PRI to an Avaya PBX. The Asterisk server has 45 PAP2T and 45 SPA-3102 devices connected via the Internet. The Asterisk server is behind a Fortinet firewall and has all necessary ports redirected to it. By itself, everything is working. I can make and receive calls to all SIP devices, check voicemail and any other service I configure on the Asterisk server. I have the relevant parts of NAT configured like externip, localnet, nat=yes and canreinvite=no. The problem only presents itself when a SIP device is trying to call an extension connected to the Avaya. Since localnet=192.168.2.0/255.255.255.0 is defined and the Fortinet firewall rewrites the source IP as its own 192.168.2.1, I think this may be the cause of my problems but why only when calling the Avaya and not other SIP extensions or Asterisk services? Since the SPA3102 has Symmetric RTP it works fine. The PAP2T on the other hand gives one way audio when you call any extension on the Avaya. The only way I can get the PAP2T to work is to change the localnet to something else then it works properly but the SPA does not. Any call I make from the SPA hangs up after a minute or so and any call I make rings the SPA but I do not get any audio. What is the proper NAT setup for something like this? Is it even possible to work with this type of NAT? Any comment would be truly appreciated. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] How to test dialplan w/o a trunk
Roderick A. Anderson wrote:
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
the LAN behind the firewall.
Can I, and if so how do I, set-up sip.conf to force my soft-phone to go
to a specific context when I take it off-hook? (The [Dial/Answer] button
in ZoIPer). Or should I set up an extension that just goes to the context?
I guessing
[613]
...
context=incoming
...
should do it.
I don't have the system on the bench yet but would like to get the
dialplan fairly close the first time. :-)
TIA,
Rod
Yeah, that should work for sip.conf after filling in the blanks, then in
extensions.conf you need an incoming context to do something like the
echo test or whatever.
Thanks,
Steve Totaro
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Re: [asterisk-users] More one way audio...
I have never seen a SIP aware firewall work with localnet and externip/externhost. You should try either disabling the SIP fixup on your firewall or remove the localnet/externip from sip.conf. Carlos Chavez wrote: I am a bit desperate trying to solve this problem. Sorry if I am abusing the list a bit with the same king of question. The problem I am having is very specific which is why it is very difficult to diagnose and fix. Basically an Asterisk server is connected via E1 PRI to an Avaya PBX. The Asterisk server has 45 PAP2T and 45 SPA-3102 devices connected via the Internet. The Asterisk server is behind a Fortinet firewall and has all necessary ports redirected to it. By itself, everything is working. I can make and receive calls to all SIP devices, check voicemail and any other service I configure on the Asterisk server. I have the relevant parts of NAT configured like externip, localnet, nat=yes and canreinvite=no. The problem only presents itself when a SIP device is trying to call an extension connected to the Avaya. Since localnet=192.168.2.0/255.255.255.0 is defined and the Fortinet firewall rewrites the source IP as its own 192.168.2.1, I think this may be the cause of my problems but why only when calling the Avaya and not other SIP extensions or Asterisk services? Since the SPA3102 has Symmetric RTP it works fine. The PAP2T on the other hand gives one way audio when you call any extension on the Avaya. The only way I can get the PAP2T to work is to change the localnet to something else then it works properly but the SPA does not. Any call I make from the SPA hangs up after a minute or so and any call I make rings the SPA but I do not get any audio. What is the proper NAT setup for something like this? Is it even possible to work with this type of NAT? Any comment would be truly appreciated. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] 3U server chassis Digium TE405P?
Gordon Henderson wrote: On Tue, 13 May 2008, Steve Totaro wrote: On Tue, May 13, 2008 at 7:26 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 12 May 2008, Steve Totaro wrote: You can put a TE405 in a 1 server (horizontally of course). I've just built a bit of an experimental system with 2 PCI cards in a 1U box. Quite a neat little system. The cards are a TDM400 and a TE120P. Will they work? Well, they seem to, so-far... Be intersting to see how it behaves under load and I'll get a chance to find out in the next few days. Gordon Gordon, It may work. It's just an experiment, but who knows :) Just a benchmark from my experience, a sing core HP DL 360 @ ~3ghz and two gigs of RAM gave me ~75 CPU usage in top. This box was used simply as a PSTN to SIP (ULAW) gateway with just the required features and programs. I was using a Sangoma four port T1 card with 95 simultaneous calls (NFAS). This is for a small office of 30 people - only 10 channels of the PRI are lit. The TDM card is for outgoing calls to a 2-port Premicell unit (analogue GSM adapter with LCR to mobile phones) and for 2 fax machines - which is the only thing I'm a shade concerend about - the analogue FAX through the TDM board and back out via the PRI board.. (incoming faxes are handled by spandsp/RxFAX and sent via email) And this in only a mere 1.3GHz VIA board too. 384MB of free RAM (512 in total, but the OS lives in a 128MB ramdisk) No transcoding and everything is custom compiled/built. (Asterisk 1.2) I actually wanted them to go for a Xorcom channel bank, as they have a door opener/bell-push to manage somehow too, but they weren't keen on it, and the Xorcom is a bit over kill - if only they did a 2+2 or even 4+4 (FXO/FXS) unit rather than an 8+8 ... (or even a stand-alone IO port!) I'll let you know how it fares :) Gordon Gordon, Did you hijack someone else's thread? You should really start your own. Anyways, I have had great luck with TDM card to TDM card faxes on the same box. I think you want echocancelwhenbridged=no at least that made TDM faxing almost perfect for my installations. The only thing I would be concerned with are the sometimes painful IRQ issues. Thanks, Steve Totaro ___ -- 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] queue problem
I have a queue with the following setting. total queue member =30, autofill=1, timeout=25, monitor_format=wav49 asterisk 1.4.18 In busy hour, the loading of CPU reaches over 300%. At that moment, all members are occupied and many calls are waiting in the queue. There will be choppy and line cut at such high CPU loading. My questions: 1. What is the max capacity of a server to handle a queue in term of queue member and calls? 2. After every 25s, the call will be switched from agent to another agent. Can I do something, say execute a CLI or shell command before it switches to another agent? ango ___ -- 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] 3U server chassis Digium TE405P?
Andreas van dem Helge wrote: A quality 3U chassis will mount the cards parallel to the mainboard with the use of a riser card, just as a 1U chassis does. If you are intent on sourcing the components yourself may I suggest a Tyan or Supermicro barebones server? I think that is the best solution for integration in these sort of specialized systems. I know they've saved me many headaches. On Mon, May 12, 2008 at 11:29 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- 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 Thanks for the suggestion, I'm actually working with SuperMicro on the deal. They ended up finding a 4U chassis (they had previously stated that they didn't sell 4u's anymore, but they found one in the back of the warehouse or something along those lines), and are shipping my server today. ___ -- 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] 3U server chassis Digium TE405P?
Have you tried GetVariableCommand and GetFullVariableCommand? See http://asterisk-java.org/development/apidocs/org/asteriskjava/fastagi/co mmand/GetFullVariableCommand.html. Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Tuesday, May 13, 2008 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 3U server chassis Digium TE405P? Andreas van dem Helge wrote: A quality 3U chassis will mount the cards parallel to the mainboard with the use of a riser card, just as a 1U chassis does. If you are intent on sourcing the components yourself may I suggest a Tyan or Supermicro barebones server? I think that is the best solution for integration in these sort of specialized systems. I know they've saved me many headaches. On Mon, May 12, 2008 at 11:29 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- 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 Thanks for the suggestion, I'm actually working with SuperMicro on the deal. They ended up finding a 4U chassis (they had previously stated that they didn't sell 4u's anymore, but they found one in the back of the warehouse or something along those lines), and are shipping my server today. ___ -- 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] 3U server chassis Digium TE405P?
On Tue, 13 May 2008, Steve Totaro wrote: Gordon, Did you hijack someone else's thread? You should really start your own. Not really - I was replying to the 3U/1U thread and rambled on a bit ... Anyways, I have had great luck with TDM card to TDM card faxes on the same box. I think you want echocancelwhenbridged=no at least that made TDM faxing almost perfect for my installations. The only thing I would be concerned with are the sometimes painful IRQ issues. They're separate. Thanks, Gordon ___ -- 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 1.4.19.2 Released
The Asterisk.org development team has released Asterisk version 1.4.19.2. This release includes some IAX2 channel driver updates. Asterisk 1.4.19.1 was released to address an IAX2 security vulnerability. Unfortunately, the changes to address the security issue had an unfortunate negative impact on IAX2 performance in Asterisk. These issues have been addressed and the related fixes are included in this release. The performance of IAX2 in Asterisk due to these changes should be far better than it was even before the changes were made for the security issue. Anyone that uses IAX2 should use this release instead of 1.4.19.1. http://downloads.digium.com/pub/telephony/asterisk/ Thank you for your continued support of Asterisk! ___ -- 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] Call only for registered sip users...
What I need to configure in my * to permit make calls only registered sip users?? 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] Call only for registered sip users...
Do you mean What do I need to configure on my * installation so that only registered sip users can make calls? ? If so, you are going to need to give a lot more details regarding your current configuration for you to get any answers. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Tuesday, May 13, 2008 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call only for registered sip users... What I need to configure in my * to permit make calls only registered sip users?? 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] Call only for registered sip users...
On Tue, 13 May 2008, equis software wrote: What I need to configure in my * to permit make calls only registered sip users?? 1) RTFM 2) Google 3) http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf 4) Ask again with more detail in the body and more specificity in the subject. Thanks in advance, Steve Edwards [EMAIL PROTECTED] 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
[asterisk-users] BLF Compatible Phones
I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- 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] How to test dialplan w/o a trunk
Steve Totaro wrote:
Roderick A. Anderson wrote:
I'm working my way through the Starfish book again trying to rid myself
of the baggage ({sip, extensions, voicemail}.conf) I brought from
another system and build the dialplan I really want.
I will be doing this on a test system without a trunk. Just sitting on
the LAN behind the firewall.
Can I, and if so how do I, set-up sip.conf to force my soft-phone to go
to a specific context when I take it off-hook? (The [Dial/Answer] button
in ZoIPer). Or should I set up an extension that just goes to the context?
I guessing
[613]
...
context=incoming
...
should do it.
I don't have the system on the bench yet but would like to get the
dialplan fairly close the first time. :-)
TIA,
Rod
Doug, Steve; thanks for the reply. I'll go for the low-hanging-fruit
(sip.conf) first. If time permits I'll test both.
Again thanks to you both.
Rod
--
Yeah, that should work for sip.conf after filling in the blanks, then in
extensions.conf you need an incoming context to do something like the
echo test or whatever.
Thanks,
Steve Totaro
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Re: [asterisk-users] BLF Compatible Phones
[EMAIL PROTECTED] schrieb: I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- 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 Hello, i would say that the best phone u can get at the moment is the Linksys SPA962 with an SPA932 Expansion Modul. Not only BLF and short dial works perfect without any problem, also a pickup runs without doing to much tricks in the asterisk config. by the way the Spa942 doesnt have blf. Maybe this one could be interesting for u too (www.snom.com) the Snom360 but it has some bugs, but all of the one u said have less or more :) best regards steve smith ___ -- 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] BLF Compatible Phones
We use the linksys 942's and they work flawlessly and are easy to setup The CISCO phones do not come with SIP, you have to upgrade their firmware from a TFTP server. [EMAIL PROTECTED] wrote: I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- 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] BLF Compatible Phones
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is that they support a max of 50 BLF subscriptions... you can setup up to 180 blf keys with 3 560Ms but it will still only subscribe to a max of 50... from what I understand it's a firmware limitation. For 4-6 phones you could probably get away with doing it directly on the 57i with no 560M's (or 536M's) too many more phones and you'd need the sidecars just for the extra buttons I think. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 13, 2008 3:50 PM To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: [asterisk-users] BLF Compatible Phones I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- 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] BLF Compatible Phones
One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Hi Dayton, It's even easier than that. With an asterisk PBX your receptionist shouldn't be picking up analog lines directly for either incoming or outgoing calls. S/he will be able to sit back and file their nails and answer the phone when it rings, and to dial out just let asterisk manage the lines for that as well. You don't have to worry about anyone managing the lines, let the software do the work. If you fill up all the lines with either incoming or outgoing calls you'll run into issues of course, but then it's a simple case of adding lines to meet demand. You will want to keep track of usage for this purpose, but it need not be actively managed by the receptionist unless you specifically want them to. A small script is more reliable and better for most cases. If you've already got the server set up at home, dig through ATFOT a little more and start configuring a line or two at home and you'll start to realize how the config works. Best regards, David [EMAIL PROTECTED] ___ -- 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] Call retard from a softphone to a hardphone
Hi group I'm newbie on Asterisk so I followed the Linux Networking CookBook by Carla Schroder to make my first call. My asterisk box is on a Debian box with an public static IP. The clients (2) are with dynamic private IP's I'm using SJphone on a PC and a Linksys PAP2-NA to make calls between them. Both of them register well on my Asterisk server but when I call from the SJPhone to the PAP2 the voice comes with retard, and progressively the voice is bad. This is my sip.conf [general] context=default port=5060 bindaddr=0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw ;SARAHC is the PAP2 [sarahc] ;Sarah Connor type=friend username=sarahc secret=5656 host=dynamic disallow=all allow=alaw allow=ulaw dtmfmode=rfc2833 outgoinglimit=1 context=local-users ;DUTCHS is the SJPhone [dutchs] ;Dutch Schaeffer type=friend username=dutchs secret=6767 host=dynamic context=local-users Sorry if I'm not giving enough information because I'm new to this wonderful tool but any idea or guide would be very good. Thanks in advanced Carlos Bernat ___ -- 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] Installation Question
I am about to start my first Asterisk installation and will only be using IP phones (either Snom's or Aastra's), I have a local voip Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this up inside Xen on a CentOS pv guest. I understand from reading old posts since I am not needing any hardware peripheral cards that this should be acceptable. Can I omit the Zaptel and Libpri portions of the install in lieu of not needing the hardware support? Thanks! jlc ___ -- 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-biz] ANI
Bill Michaelson wrote: Alex Balashov wrote: Steve Totaro wrote: This make more sense: Open WiFi AP (or cracked WEP) hacked Asterisk box (who sets the CID/ANI Telco -- terminated to the PSTN Well, sure, but you can do far worse things than spoof ANI/CID with that kind of mischief. The sort of things generated in the scenario you described are hard to track down whether they're telephony-related or not. Precisely right, and in the general case, it seems that the essential problem is the lack of general awareness that certain forms of identification are unreliable. Thus the perceived need to clear the innocent. And also, perhaps, the reason for excessive apathy about the (general) problem in many corners. Referring back to my earlier suggestion about public key authentication, a more widespread appreciation and understanding of it's applicability in various realms would go a long way toward helping solve many problems ranging from spam and phishing to stuff like this. It's a mind-share/social problem. There is nothing inherently wrong with spoofing; the problems arise when the receiver is unduly deceived. I motion that this thread be moved to the Asterisk Users (already copied to Users List) For those that do not subscribe to the Biz list, this thread may be interesting to you. http://lists.digium.com/pipermail/asterisk-biz/2008-May/subject.html I am done giving examples of what could be done as far as current exploits. The purpose was to clue some people into what can actually be done that could cause *real harm*. I would like to see what Bill and others can offer as solutions. This particular issue could result in many forms of real harm and is worth more discussion. *Maybe the Asterisk Community can do more than talk about Asterisk. We are numerous, smart, and many are influential or have influential contacts.* Thanks, Steve Totaro ___ -- 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] Installation Question
On Tue, May 13, 2008 at 4:56 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: I am about to start my first Asterisk installation and will only be using IP phones (either Snom's or Aastra's), I have a local voip Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this up inside Xen on a CentOS pv guest. I understand from reading old posts since I am not needing any hardware peripheral cards that this should be acceptable. Can I omit the Zaptel and Libpri portions of the install in lieu of not needing the hardware support? Thanks! jlc Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro ___ -- 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] Installation Question
Joseph L. Casale wrote: I am about to start my first Asterisk installation and will only be using IP phones (either Snom's or Aastra's), I have a local voip Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this up inside Xen on a CentOS pv guest. I understand from reading old posts since I am not needing any hardware peripheral cards that this should be acceptable. Can I omit the Zaptel and Libpri portions of the install in lieu of not needing the hardware support? Correct. But you will need ztdummy if you want to use MeetMe() conferencing, which is part of zaptel. However, libpri would still not be required. If you don't care about conferencing, no zaptel needed. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Installation Question
Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro Hi Steve, I read the wiki and see this provides timing for Asterisk. Can you point me toward a description of what exactly this does? I was checking out the tutorial at http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm and noticed they never compiled either this or the Libpri which is what prompted me to assume I may not need it in my scenario. Appreciate the help! jlc ___ -- 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] queue problem
On Wed, May 14, 2008 at 01:27:38AM +0800, Rilawich Ango wrote: I have a queue with the following setting. total queue member =30, autofill=1, timeout=25, monitor_format=wav49 asterisk 1.4.18 In busy hour, the loading of CPU reaches over 300%. At that moment, all members are occupied and many calls are waiting in the queue. There will be choppy and line cut at such high CPU loading. Hi, I was having huge problems with AsteriskNow 1.0.1 which is packaged with asterisk 1.4.18(.1? not sure). Most of them came with the deploiment of our support center call queue. With only 2/3 agents and max 6/10 simultaneous calls the system goes wazaa and eat the 4 cores of the xeon 1.6 cpu, users gets stucks in 'in use' state (while normaly this flags doesn't work) and everything goes from bad to worse. I've rebuild it using etch/amd64 and manual build of asterisk 1.4.20rc2 (1.4.19.2 wasn't out and i have some IAX calls) and now everything is fine. My questions: 1. What is the max capacity of a server to handle a queue in term of queue member and calls? Do you use IAX2 ? there is major improvement in this with 1.4.19.2 / 1.4.20. 2. After every 25s, the call will be switched from agent to another agent. Can I do something, say execute a CLI or shell command before it switches to another agent? -- Benoit ___ -- 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] Installation Question
On Tue, 13 May 2008, Alex Balashov wrote: Joseph L. Casale wrote: I am about to start my first Asterisk installation and will only be using IP phones (either Snom's or Aastra's), I have a local voip Supplier provisioning 1/2 a dozen did's for me. I am hoping to set this up inside Xen on a CentOS pv guest. I understand from reading old posts since I am not needing any hardware peripheral cards that this should be acceptable. Can I omit the Zaptel and Libpri portions of the install in lieu of not needing the hardware support? Correct. But you will need ztdummy if you want to use MeetMe() conferencing, which is part of zaptel. However, libpri would still not be required. If you don't care about conferencing, no zaptel needed. And MOH? Thanks in advance, Steve Edwards [EMAIL PROTECTED] 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] BLF Compatible Phones
SPA942s do not currently support BLF keys. The four lit buttons are line keys only with the current firmware, although our Linksys rep has assured us that it's a feature to be supported soon. John Signorello wrote: We use the linksys 942's and they work flawlessly and are easy to setup The CISCO phones do not come with SIP, you have to upgrade their firmware from a TFTP server. ___ -- 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] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
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[asterisk-users] Zaptel Install Error
I was just following the wiki and installing Zaptel under my CentOS 5.1 Xen DomU and after starting the service, the vm crashed. Now when restarting it, I get the following. Any ideas? Thanks! jlc Kernel BUG at kernel/timer.c:331 invalid opcode: [1] SMP last sysfs file: /class/zaptel/zapctl/dev CPU 0 Modules linked in: ztdummy(U) xpp_usb(U) xpp(U) wcusb(U) wctdm(U) wcfxo(U) wctdm24xxp(U) wcte11xp(U) wct1xxp(U) wcte12xp(U) wct4xxp(U) tor2(U) zaptel(U) crc_ccitt dm_multipath parport_pc lp parport pcspkr dm_snapshot dm_zero dm_mirror dm_mod xenblk ext3 jbd ehci_hcd ohci_hcd uhci_hcd Pid: 853, comm: modprobe Not tainted 2.6.18-53.1.19.el5xen #1 RIP: e030:[8021c055] [8021c055] __mod_timer+0x19/0xbe RSP: e02b:88000dbadd18 EFLAGS: 00010046 RAX: RBX: 806992a0 RCX: fffef0d7 RDX: 00fa RSI: fffef0dc RDI: 806992a0 RBP: 88000de5c000 R08: R09: 0020 R10: R11: R12: 88000d3ed000 R13: fffef0dc R14: 882a5580 R15: c2077940 FS: 2aabe240() GS:80599000() knlGS: CS: e033 DS: ES: Process modprobe (pid: 853, threadinfo 88000dbac000, task 88000f8d4100) Stack: 88000d3ed000 88000d3ed470 88000de5c000 88000d3ed000 88000d3ed470 882a5580 803847a7 88000dbaddef 0400 Call Trace: [803847a7] rtc_do_ioctl+0x1c5/0x701 [88121aab] :zaptel:zt_register+0x156/0x25b [80282ca0] __cond_resched+0x1c/0x44 [8026187d] _spin_lock_irq+0x9/0x14 [8025ff21] wait_for_completion+0xa1/0xaa [882a4301] :ztdummy:init_module+0x1c4/0x24b [8028e9a9] blocking_notifier_call_chain+0x2d/0x36 [8029b716] sys_init_module+0x16a6/0x1857 [8025d102] system_call+0x86/0x8b [8025d07c] system_call+0x0/0x8b Code: 0f 0b 68 5d 21 47 80 c2 4b 01 48 8d 74 24 08 48 89 df 45 31 RIP [8021c055] __mod_timer+0x19/0xbe RSP 88000dbadd18 0Kernel panic - not syncing: Fatal exception ___ -- 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] BLF Compatible Phones
On Tue, May 13, 2008 at 3:50 PM, [EMAIL PROTECTED] wrote: I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray You select lines with a Key System, a PBX simply uses a pool of lines. You may not even need or want any BLF (Busy Lamp Field) or SLA (Shared Line Appearances) because Asterisk is a PBX (for the most part). http://en.wikipedia.org/wiki/Private_branch_exchange PBXs are differentiated from key systems in that users of key systems manually select their own outgoing lines, while PBXs select the outgoing line automatically. Hybrid systems combine features of both. Thanks, Steve Totaro ___ -- 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] Zaptel Install Error
On Tue, May 13, 2008 at 7:51 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: I was just following the wiki and installing Zaptel under my CentOS 5.1 Xen DomU and after starting the service, the vm crashed. Now when restarting it, I get the following. Any ideas? Thanks! jlc Kernel BUG at kernel/timer.c:331 invalid opcode: [1] SMP last sysfs file: /class/zaptel/zapctl/dev CPU 0 Modules linked in: ztdummy(U) xpp_usb(U) xpp(U) wcusb(U) wctdm(U) wcfxo(U) wctdm24xxp(U) wcte11xp(U) wct1xxp(U) wcte12xp(U) wct4xxp(U) tor2(U) zaptel(U) crc_ccitt dm_multipath parport_pc lp parport pcspkr dm_snapshot dm_zero dm_mirror dm_mod xenblk ext3 jbd ehci_hcd ohci_hcd uhci_hcd Pid: 853, comm: modprobe Not tainted 2.6.18-53.1.19.el5xen #1 RIP: e030:[8021c055] [8021c055] __mod_timer+0x19/0xbe RSP: e02b:88000dbadd18 EFLAGS: 00010046 RAX: RBX: 806992a0 RCX: fffef0d7 RDX: 00fa RSI: fffef0dc RDI: 806992a0 RBP: 88000de5c000 R08: R09: 0020 R10: R11: R12: 88000d3ed000 R13: fffef0dc R14: 882a5580 R15: c2077940 FS: 2aabe240() GS:80599000() knlGS: CS: e033 DS: ES: Process modprobe (pid: 853, threadinfo 88000dbac000, task 88000f8d4100) Stack: 88000d3ed000 88000d3ed470 88000de5c000 88000d3ed000 88000d3ed470 882a5580 803847a7 88000dbaddef 0400 Call Trace: [803847a7] rtc_do_ioctl+0x1c5/0x701 [88121aab] :zaptel:zt_register+0x156/0x25b [80282ca0] __cond_resched+0x1c/0x44 [8026187d] _spin_lock_irq+0x9/0x14 [8025ff21] wait_for_completion+0xa1/0xaa [882a4301] :ztdummy:init_module+0x1c4/0x24b [8028e9a9] blocking_notifier_call_chain+0x2d/0x36 [8029b716] sys_init_module+0x16a6/0x1857 [8025d102] system_call+0x86/0x8b [8025d07c] system_call+0x0/0x8b Code: 0f 0b 68 5d 21 47 80 c2 4b 01 48 8d 74 24 08 48 89 df 45 31 RIP [8021c055] __mod_timer+0x19/0xbe RSP 88000dbadd18 0Kernel panic - not syncing: Fatal exception This looks like it may be your problem. http://bugs.digium.com/view.php?id=9592 (0070069) qwell - administrator 09-06-07 17:05 Closing. The simple solution here is to just comment out the #define USE_RTC in ztdummy.c. The ztxen module does not appear to be needed. Thanks, Steve Totaro ___ -- 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] Installation Question
On Tue, 2008-05-13 at 16:24 -0600, Joseph L. Casale wrote: Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro Hi Steve, I read the wiki and see this provides timing for Asterisk. Can you point me toward a description of what exactly this does? I was checking out the tutorial at http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm and noticed they never compiled either this or the Libpri which is what prompted me to assume I may not need it in my scenario. Appreciate the help! jlc sorry for responding to a question not addressed to me, you need ztdummy for timing and you need zaptel for ztdummy, no other way around, in some systems that may be not enough and you'll ever need an analog card, ___ -- 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 smime.p7s Description: S/MIME cryptographic signature ___ -- 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-biz] ANI
Regulation, laws, and controls are NOT the answer. I like the freedom I am entitled to, even with the Patriot Act. It will be a sad, sad day when all thoughts, conversations, and transactions are logged and once logged can be a form of control rather than a form of 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
[asterisk-users] No-mobo PC for USB Drives Enclosure?
I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- 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-biz] ANI
On Tue, May 13, 2008 at 9:57 PM, Alexander Lopez [EMAIL PROTECTED] wrote: Regulation, laws, and controls are NOT the answer. I like the freedom I am entitled to, even with the Patriot Act. It will be a sad, sad day when all thoughts, conversations, and transactions are logged and once logged can be a form of control rather than a form of safety. We are closer to that sad day than you know. Approaching or already past the tipping point. Thanks, Steve Totaro ___ -- 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] voicemail not sending emails
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
To: Asterisk Users List
Subject: [asterisk-users] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
___
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Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote: I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein Is the reason for separate drives security or something else? How much data will the max size drive hold? Maybe a few of these could solve your problem? http://www.buy.com/retail/product.asp?sku=206821004adid=17070dcaid=17070 Looking for a JBOD SATA enclosure with six slots but they are way expensive. Thanks, Steve Totaro ___ -- 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] voicemail not sending emails
Are you certain Asterisk is not sending the emails, rather than them not
being received? i have had problems in the past with spam filters
rejecting the emails.
On Tue, May 13, 2008 at 4:48 PM, Roberto Milani
[EMAIL PROTECTED] wrote:
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
___
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Re: [asterisk-users] Installation Question
I thought that the point that you had to have a timing source for *. That source could be the clock off the T-1. But if you didn't have something like your T1 to provide master clocking ztdummy was something to provide the required a source for timing. Joseph L. Casale wrote: Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro Hi Steve, I read the wiki and see this provides timing for Asterisk. Can you point me toward a description of what exactly this does? I was checking out the tutorial at http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm and noticed they never compiled either this or the Libpri which is what prompted me to assume I may not need it in my scenario. Appreciate the help! jlc ___ -- 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] voicemail not sending emails
Helpful?
http://lists.digium.com/pipermail/asterisk-users/2005-April/097548.html
Thanks,
Steve Totaro
On Tue, May 13, 2008 at 10:28 PM, OCG Technical Support [EMAIL PROTECTED]
wrote:
Permissions? Try running msmtp from the asterisk account? (Assuming that
is how you have it setup)
I don't know msmtp - but is there a maillog equivalent?
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roberto Milani
Sent: May 13, 2008 7:49 PM
To: Asterisk Users List
Subject: [asterisk-users] voicemail not sending emails
Hello list users
I have a very nice installation of asterisk on a mac mini.
Everything seems to work fine, call works, vm works, even message
transfer works but asterisk doesn't send any email.
this is my voicemail.conf:
[general]
mailcmd=/opt/local/bin/msmtp -t; [EMAIL PROTECTED]
;mailcmd=cat \ /tmp/asteriskvm-mail
format=wav
attach=yes
[EMAIL PROTECTED]
emailsubject=New message from ${VM_CALLERID}
emailbody=Hi, ${VM_NAME}!\n\nYou have a new message from $
{VM_CALLERID} in mailbox ${VM_MAILBOX}.
fromstring=My Telephone System
;max and min length of a message
maxmessage = 180
maxlogins = 3
[default]
100 = 4711,Front Desk,[EMAIL PROTECTED]
as you can see I'm using msmtp for mail and I tested it outside
asterisk an it works.
from the commented line you can se that I tried to cat the output to a
file but that never happens.
It really seems that asterisk don't send the emails.
any suggestions?
Thanks
Roberto
___
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Re: [asterisk-users] Installation Question
You don't need it except for a few applications such as meetme and IAX2.. I have come to always put some sort of timing hardware in a system because ztdummy can be flaky under high use. A TDM400P with and FXS module is usually what I suggest for fax, emergency phone, or whatever else. It works great. The timing thing is a common misconception and understandably so because of the same naming. There are two different types of timing. One is RTP (I think) and the other is the PSTN T1 timing you refer to. Thanks, Steve Totaro On Tue, May 13, 2008 at 10:45 PM, Al Baker [EMAIL PROTECTED] wrote: I thought that the point that you had to have a timing source for *. That source could be the clock off the T-1. But if you didn't have something like your T1 to provide master clocking ztdummy was something to provide the required a source for timing. Joseph L. Casale wrote: Sure if you don't need ztdummy, or is there a newfangled way around that? Thanks, Steve Totaro Hi Steve, I read the wiki and see this provides timing for Asterisk. Can you point me toward a description of what exactly this does? I was checking out the tutorial at http://www.hotbutteredit.com/video/voip/asterisk/asterisk_install.htm and noticed they never compiled either this or the Libpri which is what prompted me to assume I may not need it in my scenario. Appreciate the help! jlc ___ -- 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] No-mobo PC for USB Drives Enclosure?
On Tue, 2008-05-13 at 22:46 -0400, Steve Totaro wrote: On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote: I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein Is the reason for separate drives security or something else? How much data will the max size drive hold? Maybe a few of these could solve your problem? http://www.buy.com/retail/product.asp?sku=206821004adid=17070dcaid=17070 Looking for a JBOD SATA enclosure with six slots but they are way expensive. The drives are 750GB drives, each one a different related set of apps from a different Asterisk machine. I've consolidated them all into a single Asterisk server. And I already have the existing PC chassis and power supply, as well as the $10 each SATA/USB adapters. If I can just figure out how to power them from the PC power supply without plugging in a useless motherboard, I'll have it done without spending any money (other than whatever cheap part tells the power supply to run without a mobo). Thanks, Steve Totaro -- (C) Matthew Rubenstein ___ -- 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] No-mobo PC for USB Drives Enclosure?
At 11:45 PM on 13 May 2008, Matthew Rubenstein wrote: The drives are 750GB drives, each one a different related set of apps from a different Asterisk machine. I've consolidated them all into a single Asterisk server. And I already have the existing PC chassis and power supply, as well as the $10 each SATA/USB adapters. If I can just figure out how to power them from the PC power supply without plugging in a useless motherboard, I'll have it done without spending any money (other than whatever cheap part tells the power supply to run without a mobo). What I do to power up a supply without a mobo is short the green wire to a black one (on an ATX 20-pin connector) with a small piece of metal--like a staple straightened and then bent in half, or a piece of a paper clip. As soon as you plug the supply into AC, it powers up. Not sure if this is very safe... but it works for me every time. I guess you might want to avoid letting the shunt contact the case... however, given that the black wires are ground, I wouldn't worry too much about it. Anyway, this advice comes with no warranty... Use it at your own risk. If anything breaks, you get to keep both parts. ;-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #22: Wondering which Debian mirror is best for you? Check out the apt-spy and netselect-apt packages, which can give you information about how various mirror sites perform. ___ -- 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] No-mobo PC for USB Drives Enclosure?
CAUTION: doing this could be bad, i take no responsibility etc etc Put a paper clip (or any join) between the green wire and any of the black wires on an ATX power supply main lead to power it up without a motherboard - google power up atx supply without motherboard if you don't trust me Enjoy! - Nick Matthew Rubenstein wrote: On Tue, 2008-05-13 at 22:46 -0400, Steve Totaro wrote: On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote: I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein Is the reason for separate drives security or something else? How much data will the max size drive hold? Maybe a few of these could solve your problem? http://www.buy.com/retail/product.asp?sku=206821004adid=17070dcaid=17070 Looking for a JBOD SATA enclosure with six slots but they are way expensive. The drives are 750GB drives, each one a different related set of apps from a different Asterisk machine. I've consolidated them all into a single Asterisk server. And I already have the existing PC chassis and power supply, as well as the $10 each SATA/USB adapters. If I can just figure out how to power them from the PC power supply without plugging in a useless motherboard, I'll have it done without spending any money (other than whatever cheap part tells the power supply to run without a mobo). Thanks, Steve Totaro ___ -- 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] No-mobo PC for USB Drives Enclosure?
To turn on an ATX power supply that isn't connected to a motherboard use a wire or paper clip to short the green wire (PS_ON) to any one of the black wires (COM). Pins 14 and 15 Now that's the cheapest solution I can give you Alex Snip... If I can just figure out how to power them from the PC power supply without plugging in a useless motherboard, I'll have it done without spending any money (other than whatever cheap part tells the power supply to run without a mobo). Thanks, Steve Totaro -- (C) Matthew Rubenstein ___ -- 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] No-mobo PC for USB Drives Enclosure?
If I understand right, your problem is that the power supply won't turn on ? ATX power supplies can be told to turn on by jumpering 2 pins on the motherboard power connector. From memory its the Green wire and one of the black wires, I usually use the next one inwards. Pinouts for the connector can be found via Google. If the power supply also has an external on/off switch you can jumper these pins and use the switch to turn the power on or off. Hope this helps, Col - Original Message - From: Matthew Rubenstein [EMAIL PROTECTED] To: Asterisk -Users asterisk-users@lists.digium.com Sent: Wednesday, May 14, 2008 12:22 PM Subject: [asterisk-users] No-mobo PC for USB Drives Enclosure? I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- 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 -- No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.16/1430 - Release Date: 5/13/2008 7:31 AM ___ -- 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] No-mobo PC for USB Drives Enclosure?
On Wed, 2008-05-14 at 14:06 +1000, Col Ferguson wrote: If I understand right, your problem is that the power supply won't turn on ? ATX power supplies can be told to turn on by jumpering 2 pins on the motherboard power connector. From memory its the Green wire and one of the black wires, I usually use the next one inwards. Pinouts for the connector can be found via Google. If the power supply also has an external on/off switch you can jumper these pins and use the switch to turn the power on or off. Hope this helps, Thanks, that sounds like exactly what I was looking for. Is there any safety risk from jumpering that sensor? Like some kind of extra sensor, like voltage feedback, temperature or somesuch. If this works, it might point to a good way to reduce redundant Asterisk servers, which suck power, by just plugging the drive from each old server into the USB of a single server with a merged dialplan and a few other tweaks to point at several different mounted drives, rather than one per host/IP#. Col - Original Message - From: Matthew Rubenstein [EMAIL PROTECTED] To: Asterisk -Users asterisk-users@lists.digium.com Sent: Wednesday, May 14, 2008 12:22 PM Subject: [asterisk-users] No-mobo PC for USB Drives Enclosure? I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- 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 -- No virus found in this incoming message. Checked by AVG. Version: 7.5.524 / Virus Database: 269.23.16/1430 - Release Date: 5/13/2008 7:31 AM -- (C) Matthew Rubenstein ___ -- 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] No-mobo PC for USB Drives Enclosure?
This will work: http://www.newegg.com/Product/Product.aspx?Item=N82E16899705001 I assume you have devised a way to power the USB to serial adapters from the PC power supply. FWIW I think your system is inefficient but maybe you do need 750gb per each installation. Each to his own. On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein [EMAIL PROTECTED] wrote: I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- 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
