Re: [asterisk-users] async agi question
Thanks, I was not familiar with this application. /Henrik Kevin P. Fleming skrev: Henrik Westerberg wrote: Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. The ExternalIVR interface, while not quite as feature-filled as AGI, does in fact work in a true non-blocking fashion, and supports exactly what you are looking for. In fact, needing to be able to stop playback of prompts asynchronously was the primary reason it was developed. ___ -- 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] PRI span debug out put - failing international calls
I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 07 21 80 31 32 33 34 35] Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '12345' ] [70 0e a1 30 30 33 35 33 31 36 36 30 32 33 31 31] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316602311' ] [a1] Sending Complete (len= 1) q931.c:2879 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3414 q931_receive: call 32774 on channel 1 enters state 3 (Outgoing call Proceeding) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3549 q931_receive: call 32774 on channel 1 enters state 12 (Disconnect Indication) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2715 q931_release: call 32774 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3489 q931_receive: call 32774 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ -- 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] PRI span debug out put - failing international calls
In article [EMAIL PROTECTED], Mr Gabriel [EMAIL PROTECTED] wrote: I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 07 21 80 31 32 33 34 35] Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '12345' ] This is not correct - you are presenting an internal number as a Caller-ID with a TON of National. You should set a valid Caller-ID in your dialplan before calling Dial(). Or via whatever GUI you might be using. However, this probably isn't the cause of failure - BT should just ignore the Caller-ID. [70 0e a1 30 30 33 35 33 31 36 36 30 32 33 31 31] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316602311' ] However, I think this is wrong, and probably the cause of the failure. It is saying that you have pridialplan=national (the default), but you are giving a complete number. This is effectively dialling the number '00035316602311'. The first thing to make sure is that your pridialplan= and xxxprefix= directives in zapata.conf are BEFORE the channels to which they apply. When you have a channel= directive in the file, those channels will be created with the parameters that have ALREADY been seen in the file, and any parameters that come later, won't apply to those channels. Also, don't forget you need to restart Asterisk if you change the details in zapata.conf (perhaps reload might be enough, but I'm never sure). [a1] Sending Complete (len= 1) q931.c:2879 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3414 q931_receive: call 32774 on channel 1 enters state 3 (Outgoing call Proceeding) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] This response field is telling you there is no such number as 00035316602311 [remainder snipped] Things you need to try, exactly: 1) Make sure the pri and prefix directives are before the channel list. 2) Change to pridialplan=unknown and try dialling both UK and Ireland. 3) Change to pridialplan=dynamic with nationalprefix=0 and internationalprefix=00, and try dialling both UK and Ireland. If that still doesn't work, and you are happy to give me remote ssh access, email me privately. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MedHelp 34189
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[asterisk-users] meetme problem maybe connected to features.conf
Hello. I have a strange problem with the MeetMe application. Configured is a misdn msn to go into a preconfigured MeetMe room. exten = 12,1,MeetMe(1234,pIM) The first caller gets the prompt to enter the pin and then gets connected to the MeetMe room. The second caller gets also the prompt but after entering the right key he hears a dialtone followed by the message: The number you have dialed is invalid. After that he is also connected to the MeetMe room. This behaviour vanishes if i add the q option to the MeetMe command. And more strange this behaviour also vanishes if i change the features.conf from: [featuremap] blindxfer = #1 disconnect = #0 atxfer = #2 to: [featuremap] blindxfer = *1 disconnect = *0 atxfer = *2 I'm using asterisk-1.4.22, dahdi and misdn-1.1.7.2. Someone has an idea whats going on here? best regards t. ___ -- 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] Oslec issue
Ok Joseph. Don't worry, take your time :-) For what's concerning the quality: I can assume my phone line is an exception because it has a lot of echo. I've spent a LOT of time trying to have an SPA3102 and an HT488 working without any reasonable result. I'm playing for fun with zaptel/dahdi ec's since years and I was never able to have a satisfying result with any ec provided with it. Neither the fxotune process, neither any Tx/Rx gain or echo training parameter tuning, neither Digium people that connected to my server 3 or 4 years ago were able to completely solve any echo issue. Some years ago I had the opportunity to test on my line HPEC with a customer's box equipped with a TDM400P and I was impressed by the quality. I think OSLEC is a very good piece of code. It's working fine with my line and my Grandstream phones and, the must important thing, it's open and free to use. Sometimes I can ear some echo or strange effects at the very beginning of a call, but this is something that I can accept. In the past I tried to modify the zaptel sources in order to prevent them to free the oslec instance at each call. I think that my mods were not working on systems where more than one zap channel was present and I was not able to test it on these type of situations. Thank you and bye Marco Signorini Joseph L. Casale wrote: I spent some time to understand what's missing in the OSLEC patch for dahdi... I can confirm the same problem you reported some days ago and I need OSLEC for home personal use. Wow, Appreciate the info! I will need a few days to get this done. Out of curiosity, how do you find this ec's quality compared to the shipped modules and hpec? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI span debug out put - failing international calls
- Original Message - From: Tony Mountifield [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, 8 December, 2008 10:17:33 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] PRI span debug out put - failing international calls In article [EMAIL PROTECTED], Mr Gabriel [EMAIL PROTECTED] wrote: I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 07 21 80 31 32 33 34 35] Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '12345' ] This is not correct - you are presenting an internal number as a Caller-ID with a TON of National. You should set a valid Caller-ID in your dialplan before calling Dial(). Or via whatever GUI you might be using. However, this probably isn't the cause of failure - BT should just ignore the Caller-ID. [70 0e a1 30 30 33 35 33 31 36 36 30 32 33 31 31] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316602311' ] However, I think this is wrong, and probably the cause of the failure. It is saying that you have pridialplan=national (the default), but you are giving a complete number. This is effectively dialling the number '00035316602311'. The first thing to make sure is that your pridialplan= and xxxprefix= directives in zapata.conf are BEFORE the channels to which they apply. When you have a channel= directive in the file, those channels will be created with the parameters that have ALREADY been seen in the file, and any parameters that come later, won't apply to those channels. Also, don't forget you need to restart Asterisk if you change the details in zapata.conf (perhaps reload might be enough, but I'm never sure). [a1] Sending Complete (len= 1) q931.c:2879 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3414 q931_receive: call 32774 on channel 1 enters state 3 (Outgoing call Proceeding) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] This response field is telling you there is no such number as 00035316602311 [remainder snipped] Things you need to try, exactly: 1) Make sure the pri and prefix directives are before the channel list. 2) Change to pridialplan=unknown and try dialling both UK and Ireland. 3) Change to pridialplan=dynamic with nationalprefix=0 and internationalprefix=00, and try dialling both UK and Ireland. If that still doesn't work, and you are happy to give me remote ssh access, email me privately. **Gabriel says** I have made the amendments as advised, but the issues still exists - I have attached another PRI debug of attempted international call, and also the zapata.conf - I am at a lost, because we feel that everything is actually complete - and help will be appreciated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Making new call for cr 32770 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03
Re: [asterisk-users] Echo Cancelation
Robert Boardman wrote: I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Modems and FAXes already disable echo cancellation by themselves during all setup (on the Asterisk console you'll see messages telling you the canceller was disabled due to reception of a CED tone). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] meetme problem maybe connected to features.conf
Hello. I have a strange problem with the MeetMe application. Configured is a misdn msn to go into a preconfigured MeetMe room. exten = 12,1,MeetMe(1234,pIM) The first caller gets the prompt to enter the pin and then gets connected to the MeetMe room. The second caller gets also the prompt but after entering the right key he hears a dialtone followed by the message: The number you have dialed is invalid. After that he is also connected to the MeetMe room. This behaviour vanishes if i add the q option to the MeetMe command. And more strange this behaviour also vanishes if i change the features.conf I think i posted to quick. The error has nothing to do with features.conf. But the q option really solves this issue. I'm a little bit lost. So if someone has an idea i would be really happy. cheers t. ___ -- 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] meetme problem maybe connected to features.conf
Here is my misdn.conf. [general] debug=0 tracefile=/var/log/asterisk/misdn.log bridging=yes [default] language=de nationalprefix=0 internationalprefix=00 musicclass=default overlapdial=yes senddtmf=yes hold_allowed=yes [TEports] ports=1,2,3,4 context=eingehend msns=* [NTport1] ports=5 context=alarmanlage msns=* [FAX] ports=6 context=fax-out msns=* cheers t. ___ -- 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] Anyone know which vulnerability specifically they are referring to?
http://www.networkworld.com/news/2008/120608-fbi-criminals-auto-dialing-with-hacked.html?Inform=nlnetht=rn_120808nladname=120808dailynewsamal Criminals are taking advantage of a bug in the Asterisk Internet telephony system that lets them pump out thousands of scam phone calls in an hour, the U.S. Federal Bureau of Investigation warned Friday. The FBI didn't say which versions of Asterisk were vulnerable to the bug, but it advised users to upgrade to the latest version of the software. Asterisk is an open-source product that lets users turn a Linux computer into a VoIP telephone exchange.___ -- 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 survey
just had an interesting random 'survey' call. TMC publishing are calling asterisk installers (I'm not one, but they probably got my name from the asterisk lists) asking about asterisk installations, how many installs per month, how many employees in the company etc etc Unfortunately the call center jockey didn't know which vendor was paying for he suvery but i bet someone is building a decent size list for something as I'm sure TMC dont come cheap to provide this research. Cheers, Dean Collins www.Cognation.net ___ -- 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] Possible to get Courtesy Tone on attended transfer?
Thanks for the answer Terry, it's kind of what I expected. I may have to look into using Attended transfers in Asterisk, but I think my users really prefer having the TRNSFR soft key instead of remembering a feature code. Just for fun, I added the functionality that you wanted in revision 161679 of trunk. You can get the patch by: svn diff -c 161679 http://svn.digium.com/svn/asterisk/trunk I think the patch to chan_sip.c will apply to 1.4 (the CHANGES file will obviously fail). To use, you can just add setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep to a peer in sip.conf, or set it in the dialplan. Terry ___ -- 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] Rate My Dialplan Contest Announced - Wina Phone or Copies of APSTel Visual Dialplan Std or Pro!
On Fri, 2008-12-05 at 20:55 +0200, Tzafrir Cohen wrote: Anybody with a dialplan that looks like a puppy? Reminder from a previous thread: a really silly script to graph (using gnuplot) inclusions between contexts: Even better than that... a student in one of my Asterisk classes wrote Astograph! It uses GraphViz to graph both includes and gotos between contexts. Check it out at http://projects.abourget.net/astograph/ -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording - Asterisk
Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Just trying to get this sussed out in my head! Thanks for your time. Chris Hi again, didn't get a reply to this one. I'm a bit stumped so I thought I'd try the list one more time to see if anyone has an answer. If not, thanks for reading anyway! Chris ___ -- 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] Anyone know which vulnerability specifically they are referring to?
On Monday 08 December 2008 09:11:08 am Jerry Jones wrote: http://www.networkworld.com/news/2008/120608-fbi-criminals-auto-dialing-wit h-hacked.html?Inform=nlnetht=rn_120808nladname=120808dailynewsamal Criminals are taking advantage of a bug in the Asterisk Internet telephony system that lets them pump out thousands of scam phone calls in an hour, the U.S. Federal Bureau of Investigation warned Friday. The FBI didn't say which versions of Asterisk were vulnerable to the bug, but it advised users to upgrade to the latest version of the software. Asterisk is an open-source product that lets users turn a Linux computer into a VoIP telephone exchange. Probably this one, since the summary points to that specifically: http://downloads.digium.com/pub/security/AST-2008-003.pdf -- 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
[asterisk-users] DID provider in Sweden
Anyone recommend anyone who can provide me (actually a customer, but I'm asking on their behalf) DIDs in Sweden? They already have an asterisk box (in Sweden), now want a local number for it! 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] Stability unmatched!
I never did solve my puzzle as to how to kill a Linux process that seems to be deadlocked in kernel space, but thought I would report to the list that the server did manage to stay up and continue to process several thousand calls per day: ast% uptime 11:49:37 up 1000 days, 16:30, 1 user, load average: 1.00, 1.00, 1.00 ast% Since for the past four weeks I have forced my poor users to endure infrequent audible issues associated with this event, it is now to time to finally reboot. 1000 days (almost 1001) and close to 100,000,000 calls processed! Eat that, Microsoft. j ___ -- 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] Stability unmatched!
On Mon, 8 Dec 2008, Jeff LaCoursiere wrote: I never did solve my puzzle as to how to kill a Linux process that seems to be deadlocked in kernel space, but thought I would report to the list that the server did manage to stay up and continue to process several thousand calls per day: ast% uptime 11:49:37 up 1000 days, 16:30, 1 user, load average: 1.00, 1.00, 1.00 ast% Since for the past four weeks I have forced my poor users to endure infrequent audible issues associated with this event, it is now to time to finally reboot. 1000 days (almost 1001) and close to 100,000,000 calls processed! Eat that, Microsoft. Pah! I take your 1000 days and raise you: % uptime 18:18:11 up 1146 days, 5:20, 1 user, load average: 0.08, 0.03, 0.01 Other than as a test-bed some months back, this isn't an asterisk server though. Like me, you'll likely get whinged at: OMG!!! You've not updated/patched the kernel in all that time, you're so vulnerable, etc., blah 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] Stability unmatched!
On Mon, 8 Dec 2008, Gordon Henderson wrote: Pah! I take your 1000 days and raise you: % uptime 18:18:11 up 1146 days, 5:20, 1 user, load average: 0.08, 0.03, 0.01 Other than as a test-bed some months back, this isn't an asterisk server though. No fair! Must be a server in active use for a true comparison. Curse you, anyway, for deflating my dreams of a record breaking uptime :) Like me, you'll likely get whinged at: OMG!!! You've not updated/patched the kernel in all that time, you're so vulnerable, etc., blah All of which are of course true and deserved. In fact a few months back we were actually hacked and pwned - root access achieved no less! For several days a battle of wits involving on both sides hidden directories full of RPMs and sshd's run on odd ports hiding themselves in the process list, killing processes as fast as they could be spawned, etc. In the end the intruders finally gave up and ran out of holes, at least as far as I know :):) j ___ -- 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] Stability unmatched!
The 100,000,000 calls without a crash are more impressive to me than the 1000 days of uptime. Mine crashes on crazy things like dynamic conferences, etc. :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff LaCoursiere Sent: Monday, December 08, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Stability unmatched! On Mon, 8 Dec 2008, Gordon Henderson wrote: Pah! I take your 1000 days and raise you: % uptime 18:18:11 up 1146 days, 5:20, 1 user, load average: 0.08, 0.03, 0.01 Other than as a test-bed some months back, this isn't an asterisk server though. No fair! Must be a server in active use for a true comparison. Curse you, anyway, for deflating my dreams of a record breaking uptime :) Like me, you'll likely get whinged at: OMG!!! You've not updated/patched the kernel in all that time, you're so vulnerable, etc., blah All of which are of course true and deserved. In fact a few months back we were actually hacked and pwned - root access achieved no less! For several days a battle of wits involving on both sides hidden directories full of RPMs and sshd's run on odd ports hiding themselves in the process list, killing processes as fast as they could be spawned, etc. In the end the intruders finally gave up and ran out of holes, at least as far as I know :):) j ___ -- 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] Stability unmatched!
On Mon, 8 Dec 2008, Danny Nicholas wrote: The 100,000,000 calls without a crash are more impressive to me than the 1000 days of uptime. Mine crashes on crazy things like dynamic conferences, etc. :( To be upfront the system is only running a prepaid AGI app and routing calls for post-paid customers. Can't vouch for more complex setups - all other full fledged PBX systems I have installed go down for other reasons like power. This box has been hosted by The Planet in Dallas, which I cannot recommend enough (no affiliation - just a happy customer). It is using the full suite of protocols (IAX, SIP, H323) and doing a lot of codec translation though. j ___ -- 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] Stability unmatched!
On Mon, Dec 8, 2008 at 3:06 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Danny Nicholas wrote: The 100,000,000 calls without a crash are more impressive to me than the 1000 days of uptime. Mine crashes on crazy things like dynamic conferences, etc. :( To be upfront the system is only running a prepaid AGI app and routing calls for post-paid customers. Can't vouch for more complex setups - all other full fledged PBX systems I have installed go down for other reasons like power. This box has been hosted by The Planet in Dallas, which I cannot recommend enough (no affiliation - just a happy customer). That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stability unmatched!
On Mon, 8 Dec 2008, Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ The explosion was apparently in Houston... ___ -- 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 and FreePBX
I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 ---rw- 1 asterisk asterisk274 Dec 8 12:45 msg0003.txt 88 -- 1 asterisk asterisk 89324 Dec 8 12:45 msg0003.wav 12 -- 1 asterisk asterisk 9420 Dec 8 13:03 msg0004.WAV 12 -- 1 asterisk asterisk 9504 Dec 8 13:03 msg0004.gsm 4 ---rw- 1 asterisk asterisk268 Dec 8 13:03 msg0004.txt 92 -- 1 asterisk asterisk 92204 Dec 8 13:03 msg0004.wav 4 -- 1 asterisk asterisk 3895 Dec 8 13:19 msg0005.WAV 4 -- 1 asterisk asterisk 3894 Dec 8 13:19 msg0005.gsm 4 ---rw- 1 asterisk asterisk271 Dec 8 13:19 msg0005.txt 40 -- 1 asterisk asterisk 37804 Dec 8 13:19 msg0005.wav Asterisk runs as user asterisk as does Freepbx and Apache. Any ideas on why voicemail is not going out via email? -- 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] Stability unmatched!
Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ That's what happens when illegal aliens, er, Undocumented Americans, do all your contracting work. I saw a nasty electrical fire at a shopping center where illegals accidentally hit a clearly marked underground conduit running through a parking lot during a demo of a part of a building as part of a renovation. The electrical switching room no longer existed after the explosion/fire that followed, even the telephone pole with the cans on it burnt to a crisp. The insurance company wasn't too pleased with Undocumented Americans working on the site and initially refused to pay. They did pay but ended up putting the general contractor out of business in the end after the lawsuits. It was one of the most impressive events I've ever seen. -Ron ___ -- 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] Stability unmatched!
On Mon, 8 Dec 2008, RE Kushner List Account wrote: That's what happens when illegal aliens, er, Undocumented Americans, do all your contracting work. But they taste like chicken! :) ___ -- 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 and FreePBX
Looks like someone messed with a umask. I would guess that freepbx can't read non-permissioned files. The umask for msg0005.txt was 0770. The umask for msg0005.gsm was 0777. I'd check the shell commands that run in or under both processes for changes. BTW, you should probably be running something like Tripwire to pickup changes like this when they happen instead of when the symptoms show (That's what I'm told). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Monday, December 08, 2008 2:37 PM To: Asterisk Subject: [asterisk-users] Voicemail and FreePBX I have a customer running Asterisk 1.4.22 and FreePBX 2.5.0 that is having problems with Voicemail. They can listen to their voicemail but on the weekend it stopped delivering messages via email. The only thing I can notice is that the permissions for the files on teh voicemail directories are created with no permissions at all. Here is the listing on one of the mailboxes: 4 ---rw- 1 asterisk asterisk274 Dec 8 12:45 msg0003.txt 88 -- 1 asterisk asterisk 89324 Dec 8 12:45 msg0003.wav 12 -- 1 asterisk asterisk 9420 Dec 8 13:03 msg0004.WAV 12 -- 1 asterisk asterisk 9504 Dec 8 13:03 msg0004.gsm 4 ---rw- 1 asterisk asterisk268 Dec 8 13:03 msg0004.txt 92 -- 1 asterisk asterisk 92204 Dec 8 13:03 msg0004.wav 4 -- 1 asterisk asterisk 3895 Dec 8 13:19 msg0005.WAV 4 -- 1 asterisk asterisk 3894 Dec 8 13:19 msg0005.gsm 4 ---rw- 1 asterisk asterisk271 Dec 8 13:19 msg0005.txt 40 -- 1 asterisk asterisk 37804 Dec 8 13:19 msg0005.wav Asterisk runs as user asterisk as does Freepbx and Apache. Any ideas on why voicemail is not going out via email? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need local upstate ny asterisk tech
if you are not local, pls do not reply local is an area defined by syracuse, rochester, ithaca, and binghamton new york its an ongoing consulting gig thx _ Send e-mail faster without improving your typing skills. http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_speed_122008___ -- 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] CDR Design
Steve Murphy wrote: Well, read my draft RFC, and see if I'm on the right track. Tune into CDR Design in the subject line in this email list, and let's toss this around and see if consensus is possible. murf First my apologies for this repost, my system date got messed up and this post looked like it was sent on Nov 5th :). One of the problems I generally have with cdr's in my multi-tenant hosted VOIP world, is that the src is inexorably tied to the callerid field, this makes it a pain when you have a billing system based on TDM billing systems that have not only a src field but an originating src field. This is what allows you to know what number placed the call but still allow things like no callerid. In a perfect world the fields would be src_cid src exten. When calls do not originate from within your dialplan (external to internal calls) most of these fields would be null or repetitive. I know many of you would say you know the originator of the call by the channel, but in a multitenant situation you can't have two sip devices named [100] so you use special ID's and have to do post-processing to determine that information. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] 'dialer' application to trigger call between hardphone and number
Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via hotkey, or a right-click menu or by clicking a system tray icon. There are scads of outlook-only options (no thanks), and I've found and tried the Asterisk Dialer 1.0, which I don't like because it depends on Yahoo widgets (heavy) AND it requires nearly as many discreet actions to dial a number as just typing them on the phone itself. Ideal would be something very 'efficient' with at most two or three discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!) Any ideas? Any Happy customers? ___ -- 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] Stability unmatched!
On Mon, Dec 8, 2008 at 3:37 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ The explosion was apparently in Houston... Oops! I don't know how I got that confused... Shame on me, and I was just in Texas last week! My apologies. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dialer' application to trigger call betweenhardphone and number
This sounds like a job for a VB.NET programmer. The program would run like a DDE server and ftp a call file to your asterisk server on the desired action. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl Fife Sent: Monday, December 08, 2008 3:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 'dialer' application to trigger call betweenhardphone and number Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via hotkey, or a right-click menu or by clicking a system tray icon. There are scads of outlook-only options (no thanks), and I've found and tried the Asterisk Dialer 1.0, which I don't like because it depends on Yahoo widgets (heavy) AND it requires nearly as many discreet actions to dial a number as just typing them on the phone itself. Ideal would be something very 'efficient' with at most two or three discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!) Any ideas? Any Happy customers? ___ -- 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] 'dialer' application to trigger call betweenhardphone and number
There are a few web-based ones - is that an option at all? PaulH Danny Nicholas wrote: This sounds like a job for a VB.NET programmer. The program would run like a DDE server and ftp a call file to your asterisk server on the desired action. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Karl Fife *Sent:* Monday, December 08, 2008 3:04 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] 'dialer' application to trigger call betweenhardphone and number Does anyone know of a small lightweight windows 'dialer' application I can use to trigger a call (via call file or AMI) from any application? (The call would be placed between the target number, and the preconfigured DN of the hardphone at the user's desk) Ideally a phone number would be 'selected' from within any windows application and the call would be triggered via hotkey, or a right-click menu or by clicking a system tray icon. There are scads of outlook-only options (no thanks), and I've found and tried the Asterisk Dialer 1.0, which I don't like because it depends on Yahoo widgets (heavy) AND it requires nearly as many discreet actions to dial a number as just typing them on the phone itself. Ideal would be something very 'efficient' with at most two or three discreet actions needed to dial-- (i.e. 1:Select, 2:Hotkey--done!) Any ideas? Any Happy customers? ___ -- 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] Call Recording - Asterisk
Chris Rowson wrote: I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Chris, Make sure that all of your SIP clients are set to canreinvite=no in sip.conf. The default is canreinvite=yes, which allows RTP to bypass Asterisk. Certain things (codec translation, playback of audio files, etc.) require Asterisk to be in the RTP path, which may explain why you're recording some of the calls. If you're still missing calls, make sure Oreka is configured properly in config.xml. In particular, the AllowedIpRanges and BlockedIpRanges settings provide IP address filtering at the Oreka level. In general, I've had to configure these to prevent getting two recordings of each call (since Asterisk acts as a B2BUA) but your configuration may be too strict. Running tcpdump/Wireshark on the Oreka server will let you see exactly what's being mirrored. There is even a setting in Oreka named PcapFile that will let you playback the packet capture file over and over until you're satisfied with your configuration. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] IC3/FBI security announcement - your help needed
On Friday, the IC3 (FBI/NW3C/BJA) put out a security advisory on their website that contained a fairly vaguely worded warning about Asterisk systems being compromised and then being used as vishing (voice phishing) platforms. They were non-specific on the threat other than to advocate upgrading to newer versions of Asterisk. This announcement was done on Friday late afternoon, just as everyone was leaving for the weekend, which left us leaving frantic messages with various IC3 voicemail system deadends and emails to generic-sounding accounts. The delay in any authoritative information from IC3 quickly created a guessing game in the blogger and press community as to what was exactly the vulnerability and what were the details of this threat. The speculation here at Digium was that this was just a re-statement of an older bug from earlier this year, or it could have been entirely unrelated to Asterisk and just been a case of mis-diagnosis of poor password control. It turns out that we were correct on our first guess: this is not a new problem, and furthermore is a difficult vulnerability to exploit even on those systems that are unpatched - it would require fairly purposeful configuration to expose the system to a vishing abuse method, so it is probably the case that this was a very isolated event. We spoke with IC3 agents earlier today, and they have updated the alert to contain the correct warning (AST-2008-003) which was their original intent. There is a more complete description of the incident on the Digium blog site: http://blogs.digium.com/2008/12/06/sip-security-and-asterisk/ Other links: AST-2008-003 - http://www.asterisk.org/node/48466 Revised IC3 announcement - http://www.ic3.gov/media/2008/081205-2.aspx WHAT YOU CAN DO: Unfortunately, the news of security risks spreads faster than the news of a non-issue - secure systems aren't stories so I expect it will be an uphill effort to update all the sites which copied or re- blogged the IC3 story initially. We would very much like to enlist the community to have you try to post where you can the link to the Digium blog above - it would help keep misperceptions from becoming part of the permanent data landscape as things get slowly archived into Google-able snippets. Post in the Comments sections of any blogs you see linking to this story, or put your own $.02 in as you see fit. We'd like to keep good relations with the IC3 and FBI, and we understand how this kind of mistake can happen (even though we're uncomfortable with the results) so please set your flamethrowers on warm instead of scorch if you choose to weigh in on the topic yourself. If anyone has questions regarding this issue, please feel free to contact me via email or phone to discuss. JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] Stability unmatched!
That's what happens when illegal aliens, er, Undocumented Americans, do all your contracting work. Could it be that all fires that ever happened in the US were caused by those guys ? .. I guess 1+1=5 then . On Mon, Dec 8, 2008 at 12:50 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On Mon, Dec 8, 2008 at 3:37 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Mon, 8 Dec 2008, Kristian Kielhofner wrote: That much uptime at The Planet in Dallas? I guess you're lucky: http://www.thewhir.com/marketwatch/060208_The_Planet_Explosion_Causes_Outages.cfm http://www.datacenterknowledge.com/archives/2008/06/01/explosion-at-the-planet-causes-major-outage/ The explosion was apparently in Houston... Oops! I don't know how I got that confused... Shame on me, and I was just in Texas last week! My apologies. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
Hello, 2008/12/5 [EMAIL PROTECTED] [EMAIL PROTECTED] I will publish a tutorial in the beginning of next week about how to configure Zoiper and Asterisk to do t.38 together. Zoa. Where will you publish this tuto ? Regards ___ -- 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] Stability unmatched!
Hello Friends, Sorry to hijack the thread,but i am Asterisk beginner and am facing problem with eyebeam getting registered. If i am selecting Domain with register and receive incoming calls then i am not able to get register but if i remove the tick then i am able to register with the server but the other user cannot call me as i cannot receive incoming calls. Moreover what should i select for Send Outbound Via: I selected target domain. The other user gets registered at other place and can send and receive calls. If i dont put the tick for incoming calls i get register and can make outbound calls but in that too the other user can listen my voice but i cannot listen his voice. Is that a problem with voice settings in my laptop. The server is remotely located and has vicidialnow installed on it. Kindly solve my querry. Regards, Asterisk Beginner. On Mon, Dec 8, 2008 at 11:23 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: I never did solve my puzzle as to how to kill a Linux process that seems to be deadlocked in kernel space, but thought I would report to the list that the server did manage to stay up and continue to process several thousand calls per day: ast% uptime 11:49:37 up 1000 days, 16:30, 1 user, load average: 1.00, 1.00, 1.00 ast% Since for the past four weeks I have forced my poor users to endure infrequent audible issues associated with this event, it is now to time to finally reboot. 1000 days (almost 1001) and close to 100,000,000 calls processed! Eat that, Microsoft. j ___ -- 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] Stability unmatched!
On Tue, Dec 09, 2008 at 12:45:47PM +0530, amit mehta wrote: Hello Friends, Sorry to hijack the thread, So don't. Post a new message to the list rather than replying to an existing one. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic loading changed in asterisk 1.4
Hi, * *I found a similar problem to mine with regards to the res_config_mysql module on 1.4with regards to the problem of: *asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_prepaid.so: undefined symbol: mysql_init * from another user from an asterisk-users post which can be found at http://readlist.com/lists/lists.digium.com/asterisk-users/8/41278.html. Should I be opening a bug report? Thanks again, Mosiuoa ___ -- 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
