Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?
Thanks for your reply Jay. You're quite right. It would have been better to say npa/nxx is a relatively evenly distributed keyspace. To answer your question specifically, no, I have not done a distribution analysys (though it would be interesting). The point I was trying to make is that a serial list of all active unique 'npanxx' strings in the NANP would be an easier keyspace to index than the other 'organic' keyspaces that databases often have to work with, (such as the customer name index) -K On Fri, 15 Aug 2008 12:27:52 -0400, Jay R. Ashworth [EMAIL PROTECTED] said: On Fri, Aug 15, 2008 at 12:56:49AM -0500, Karl Fife wrote: The key-space is ideal. It's just npa/nxx lookups so it's UNIQUE and EVENLY DISTRIBUTED Based on my knowledge of the NPA/NXX space, I wouldn't expect that either a) A given batch of random DNs would have either or both NPA/NXX components evenly distributed over all the valid NPA/NXXs in the NANPA, or b) that the assigned NPA/NXXs in the NANPA are themselves evenly distributed over all the valid NPA/NXXs. Could you clarify the background that brings you to that assumption? Do you have empirical data? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running asterisk as non root user
Hi, I've followed instructions of the book AsteriskFutureOf TelephonySecEdit on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1: 2: 3: 4: 5: 6: 7: 8: 9: 10: 11: 12: [EMAIL PROTECTED] run]# /etc/init.d/asterisk restartShutting down asterisk: [FAILED]Starting asterisk: [ OK [EMAIL PROTECTED] run]# Asterisk ended with exit status 1Asterisk died with code 1.cat: /var/run/asterisk.pid: No such file or directoryAutomatically restarting Asterisk.mpg123: no process killedAsterisk ended with exit status 1Asterisk died with code 1.cat: /var/run/asterisk.pid: No such file or directoryAutomatically restarting Asterisk. The suggestion to do the following didn't work...: Edit the [directories] section of asterisk.conf and change the line that reads astrundir = /var/run TO: astrundir = /var/run/asterisk Then: mkdir /var/run/asterisk chown theuser /var/run/asterisk Edit /etc/init.d/asterisk And make sure there are no references to /var/run/asterisk.pid you want /var/run/asterisk/asterisk.pid instead Any help most welcome___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running asterisk as non root user
Shaun Wingrin wrote: Hi, I've followed instructions of the book AsteriskFutureOf TelephonySecEdit on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . snip / That sounds like it's probably a permissions thing. I wrote up a howto Asterisk as non-root on my blog here: http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ Make sure you check the init script and the permissions of the various directories that asterisk needs read/write too. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan reload and dropped calls
On Sat, Aug 16, 2008 at 2:57 PM, Todd Fulton [EMAIL PROTECTED] wrote: Hi All, This is just a basic question . . . but if I'm doing a CLI dialplan reload while in operation, do I risk dropping calls or affecting calls in progress? Nope. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pollmailboxes
1.6 UPGRADE.txt: * If you use any interface for modifying voicemail aside from the built in dialplan applications, then the option pollmailboxes *must* be set in voicemail.conf for message waiting indication (MWI) to work properly. This is because Voicemail notification is now event based instead of polling based. The channel drivers are no longer responsible for constantly manually checking mailboxes for changes so that they can send MWI information to users. Examples of situations that would require this option are web interfaces to voicemail or an email client in the case of using IMAP storage. I vote for a truly event based solution: - A web GUI modifies a mailbox (e.g. deletes a message) - The GUI triggers the polling of this specific mailbox Action: PollMailbox Mailbox: 1234 ... or Action: MailboxModifiedExternally Context: default Mailbox: 1234 Folder: Work File: msg0002 ... Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6 call-limit
1.6 UPGRADDE.txt: * SIP: The call-limit option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. One of the main uses of call-limit was actually not to enforce a call limit but to make device states work. So an explanation in UPGRADE.txt wouldn't hurt, e.g. call-limit is not needed any longer for the BLF feature. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 42
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will be on vacation until Tuesday 19th of August with limited access to voice and e-mail. If you have any urgent requests, please contact Wildix srl at 0039 0461 74 30 891 or [EMAIL PROTECTED], otherwise I will answer to your e-mail on my return. Dimitri Osler ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk -n switch
man asterisk (1.6.0-beta9) says: -n Disable ANSI colors even on terminals capable of displaying them. But on the CLI (asterisk -n -r) any core show application or core show function leaks color escape sequences. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
This is weird. Any help you can offer would be appreciated. We spent 6 hours on phone with Digium support yesterday and could not locate an issue within asterisk itself. Have you tried putting a soft phone on the same machine as the asterisk box? Put a sound card in there, connect a microphone, or at least a speaker, and try calling your soft phone. Does the call go through between the system and itself? Am I the only person who has suggested that? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running asterisk as non root user
Check this one out... http://www.voip-info.org/wiki/view/Asterisk+non-root From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin Sent: Sunday, 17 August 2008 6:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Running asterisk as non root user Hi, I've followed instructions of the book AsteriskFutureOf TelephonySecEdit on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1: 2: 3: 4: 5: 6: 7: 8: 9: 10: 11: 12: [EMAIL PROTECTED] run]# /etc/init.d/asterisk restart Shutting down asterisk: [FAILED] Starting asterisk: [ OK ] [EMAIL PROTECTED] run]# Asterisk ended with exit status 1 Asterisk died with code 1. cat: /var/run/asterisk.pid: No such file or directory Automatically restarting Asterisk. mpg123: no process killed Asterisk ended with exit status 1 Asterisk died with code 1. cat: /var/run/asterisk.pid: No such file or directory Automatically restarting Asterisk. The suggestion to do the following didn't work...: Edit the [directories] section of asterisk.conf and change the line that reads astrundir = /var/run TO: astrundir = /var/run/asterisk Then: mkdir /var/run/asterisk chown theuser /var/run/asterisk Edit /etc/init.d/asterisk And make sure there are no references to /var/run/asterisk.pid you want /var/run/asterisk/asterisk.pid instead Any help most welcome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pollmailboxes
On Aug 17, 2008, at 11:07 AM, Philipp Kempgen wrote: 1.6 UPGRADE.txt: * If you use any interface for modifying voicemail aside from the built in dialplan applications, then the option pollmailboxes *must* be set in voicemail.conf for message waiting indication (MWI) to work properly. This is because Voicemail notification is now event based instead of polling based. The channel drivers are no longer responsible for constantly manually checking mailboxes for changes so that they can send MWI information to users. Examples of situations that would require this option are web interfaces to voicemail or an email client in the case of using IMAP storage. I vote for a truly event based solution: - A web GUI modifies a mailbox (e.g. deletes a message) - The GUI triggers the polling of this specific mailbox Action: PollMailbox Mailbox: 1234 ... or Action: MailboxModifiedExternally Context: default Mailbox: 1234 Folder: Work File: msg0002 I agree that it would be a nice addition. I'm not so sure that it would remove the need to be able to enable periodic polling in Asterisk, though. For example, take IMAP storage of voicemail. There would not be any easy way to trigger the poll after someone has made changes using their IMAP client. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?
On Sun, Aug 17, 2008 at 7:10 PM, David Backeberg [EMAIL PROTECTED] wrote: This is weird. Any help you can offer would be appreciated. We spent 6 hours on phone with Digium support yesterday and could not locate an issue within asterisk itself. Have you tried putting a soft phone on the same machine as the asterisk box? Put a sound card in there, connect a microphone, or at least a speaker, and try calling your soft phone. Does the call go through between the system and itself? Am I the only person who has suggested that? At this point, if I were you, I would get a different server, an HP, IBM, SuperMicro, or something that is known to work well with Asterisk. After how many hours of troubleshooting do you give up because you are wasting more time/money than it would cost to buy a new box and be done with it? If it is hardware, I give up after about eight hours or less. Thanks, Steve Totaro Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing
Hi All, Hope someone can help. Asterisk version 1.4.14 is running and just installed zaptel-1.4.11with only ztdummy selected in menuselect . ztdummy seems to be running (as below) but still get the above error, even though I've stopped and restarted asterisk... Do I need to set ZAP_TIMING=-I ? Where do I do that? lsmod | grep ztdummy ztdummy 9256 0 zaptel190852 1 ztdummy Thanks, Shaun Wingrin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On Sat, Aug 16, 2008 at 09:35:10PM -0400, Ron Joffe wrote: On Saturday 16 August 2008 14:37, Jay R. Ashworth wrote: TBCT is a feature of LEC/IXC edge switches; there isn't much use for it in any other context. I don't care if you're using Asterisk to be an edge switch, but it's a *carrier* feature, by and large. Certainly in the specific instance I'm discussing, it is. Why would TBCT not be applicable in a scenario where * is being utilized as a slave to a main PBX. * might receive a call from the PBX, and then want to transfer it to another extension on the PBX itself. Hmmm. Perhaps. But if the Asterisk is upstream of the PBX, then the *PBX* would need to know how to deal with it. I see your point... but it's still orthogonal to what I need to know. :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pollmailboxes
Russell Bryant schrieb: On Aug 17, 2008, at 11:07 AM, Philipp Kempgen wrote: 1.6 UPGRADE.txt: * If you use any interface for modifying voicemail aside from the built in dialplan applications, then the option pollmailboxes *must* be set in voicemail.conf for message waiting indication (MWI) to work properly. This is because Voicemail notification is now event based instead of polling based. The channel drivers are no longer responsible for constantly manually checking mailboxes for changes so that they can send MWI information to users. Examples of situations that would require this option are web interfaces to voicemail or an email client in the case of using IMAP storage. I vote for a truly event based solution: - A web GUI modifies a mailbox (e.g. deletes a message) - The GUI triggers the polling of this specific mailbox Action: PollMailbox Mailbox: 1234 ... or Action: MailboxModifiedExternally Context: default Mailbox: 1234 Folder: Work File: msg0002 I agree that it would be a nice addition. I'm not so sure that it would remove the need to be able to enable periodic polling in Asterisk, though. At least in some setups (a lot actually) you would not need to poll: GUIs could definitely trigger it. For example, take IMAP storage of voicemail. There would not be any easy way to trigger the poll after someone has made changes using their IMAP client. I'm pretty sure there are IMAP servers with custom hooks (Dovecot?). Not exactly easy but doable. BTW: Does pollmailboxes _disable_ the event based notifications? UPGRADE.txt is not clear about that. Some setups might want to use a mix. Event-based with custom triggers and polling-based with pollfreq=3600 as a safety net. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing
Most likely Asterisk has been built without zaptel support. (if you built Asterisk first then zaptel second, this will happen) Rebuild Asterisk. PaulH Shaun Wingrin wrote: Hi All, Hope someone can help. Asterisk version 1.4.14 is running and just installed zaptel-1.4.11with only ztdummy selected in menuselect . ztdummy seems to be running (as below) but still get the above error, even though I've stopped and restarted asterisk... Do I need to set ZAP_TIMING=-I ? Where do I do that? lsmod | grep ztdummy ztdummy 9256 0 zaptel190852 1 ztdummy Thanks, Shaun Wingrin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users