[asterisk-users] MOH uploading is not working with 1.4
Hi am I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is not working .help me on this Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel
On 06/01/2011 05:42 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 04:10:34PM +0200, randall wrote: On 06/01/2011 03:55 PM, randall wrote: On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for another day or maybe 2 or 3 till the problem occurs again. running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel 2.6.32-5-686 i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 (happens on all 4 spans) and the following in dmesg: [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252 [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current packet[0..2]: 55 55 FC [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71 Channel 0/1, span 1 got hangup, cause 18 Is this happening in the middle of a call? Or only a while after the call ended? the bad fcs messages seem to happen random there seems to be a relation indeed, have seen them happen randomly quite spurious, but they indeed tend to happen a while after the call is made. A while after a call is made? A while after a call is ended? kept an eye on this and it seems to happen after a call is ended (+- 25 - 30 seconds) and only when dialed out, but not when another call is in progress. Maybe the provider intentionally sets layer 1 down (to save power)? sounds logical with the behaviour mentioned above That makes sense on PtMP, though I was not aware of this being used on PtP. i'm clueless on this, telco is China Unicom btw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
Are you suggesting that there are no bugs in 1.4 or 1.6? Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' Ish On Fri, 2011-06-03 at 09:54 +0530, Satish Barot wrote: Paul, With due respect to Digium work, are there no issues with Asterisk 1.8? https://issues.asterisk.org/view_all_bug_page.php [SATISH] On Thu, Jun 2, 2011 at 9:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/02/2011 10:29 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 02, 2011 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I use phone line to recive faxes?
On Thursday 02 Jun 2011, khalid touati wrote: Hi Guys, Actually My question is as in the subject, may I use a regular phone line to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8. Yes, you can. BUT, you will need some sort of FXO interface (allows the computer to connect to the telephone socket on the wall), which is supported by DAHDI (or its predecesor, Zaptel). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
If 1.8 doesn't panic for subset of PBX features for someone, you can not say it is stable. You should also look at other features and how they work with 1.8. I didn't say 1.4 or 1.6 have no bugs or issues. When there were 1.4 or 1.6.0 branches, they did have bugs. But since people started submitting bug reports, they have become quite stable. They don't get crashed as frequently as 1.8 for the same set of features(You can check it on issues.asterisk.org). When I said 'Asterisk 1.8 is not stable ENOUGH', I didn't mean 'Asterisk 1.8 is not stable AT ALL'.There are still some feature functionalities which work perfactaly on 1.4 or 1.6, create some panic on 1.8. I would consider 1.8 stable enough when anything which worked on 1.4 or 1.6, also work on 1.8. And I am optimistic about 1.8 being stable enough shortly. Let us not start a war on 1.8 stability issue. There were enough threads on 1.8 being production safe in last couple of months.Mine was just a user experience and personal view shared with somebody else. [SATISH] On Fri, Jun 3, 2011 at 1:37 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' Ish __ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voxbone numbers
I am thinking about using numbers from voxbone. Before I make up my mind if this is the right service for me I want to know what kinds of details will be found when checking up on a voxbone number. I am interested in UK numbers. Can anyone give an example on an actual voxbone number in service. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voxbone numbers
I am thinking about using numbers from voxbone. Before I make up my mind if this is the right service for me I want to know what kinds of details will be found when checking up on a voxbone number. I am interested in UK numbers. Can anyone give an example on an actual voxbone number in service. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
On Fri, Jun 3, 2011 at 11:28 AM, devr devr d...@gmx.com wrote: I am thinking about using numbers from voxbone. Before I make up my mind if this is the right service for me I want to know what kinds of details will be found when checking up on a voxbone number. I am interested in UK numbers. Can anyone give an example on an actual voxbone number in service. Try this: +44 1259340614 Temporary UK number for the VoIP Users Conference. Let me know what you see? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
Voxbone works correctly, no problem, the only problem is that you need to spend a minimum amount of 500€/month to open an account... Best regards, Olivier Le 3/06/11 11:58, randulo a écrit : On Fri, Jun 3, 2011 at 11:28 AM, devr devrd...@gmx.com wrote: I am thinking about using numbers from voxbone. Before I make up my mind if this is the right service for me I want to know what kinds of details will be found when checking up on a voxbone number. I am interested in UK numbers. Can anyone give an example on an actual voxbone number in service. Try this: +44 1259340614 Temporary UK number for the VoIP Users Conference. Let me know what you see? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
thanks for your reply This is the details I get Locality Alloa, Clackmannanshire Charging B1 National Operator Voxbone SA Locality Alloa, Clackmannanshire Charging B1 National Operator Voxbone SA My query now is willl all voxbone numbers show up as the operator as Voxbone SA as above. I wanted to find out who the service provider is on some numbers, I suspected the service to be voxbone but the operator shows as other companies. My idea on how voxbone works is that voxobone is a intermediatary enabler with the actual hardware with third parties in which case the operator will show as the hardware owner. Is this acuratrate? - Original Message - From: randulo Sent: 06/03/11 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voxbone numbers On Fri, Jun 3, 2011 at 11:28 AM, devr devr d...@gmx.com wrote: I am thinking about using numbers from voxbone. Before I make up my mind if this is the right service for me I want to know what kinds of details will be found when checking up on a voxbone number. I am interested in UK numbers. Can anyone give an example on an actual voxbone number in service. Try this: +44 1259340614 Temporary UK number for the VoIP Users Conference. Let me know what you see? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
Yesterday my 1.8 got crashed and I have nothing in log or anywhere which I can show you or submit bug. Kinda funny :( -- Sent from my iPhone On Jun 3, 2011, at 5:06 AM, Satish Barot satish4aster...@gmail.com wrote: If 1.8 doesn't panic for subset of PBX features for someone, you can not say it is stable. You should also look at other features and how they work with 1.8. I didn't say 1.4 or 1.6 have no bugs or issues. When there were 1.4 or 1.6.0 branches, they did have bugs. But since people started submitting bug reports, they have become quite stable. They don't get crashed as frequently as 1.8 for the same set of features(You can check it on issues.asterisk.org). When I said 'Asterisk 1.8 is not stable ENOUGH', I didn't mean 'Asterisk 1.8 is not stable AT ALL'.There are still some feature functionalities which work perfactaly on 1.4 or 1.6, create some panic on 1.8. I would consider 1.8 stable enough when anything which worked on 1.4 or 1.6, also work on 1.8. And I am optimistic about 1.8 being stable enough shortly. Let us not start a war on 1.8 stability issue. There were enough threads on 1.8 being production safe in last couple of months.Mine was just a user experience and personal view shared with somebody else. [SATISH] On Fri, Jun 3, 2011 at 1:37 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' Ish __ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
On 3 Jun 2011, at 11:57, devr devr wrote: My query now is willl all voxbone numbers show up as the operator as Voxbone SA as above. I wanted to find out who the service provider is on some numbers, I suspected the service to be voxbone but the operator shows as other companies. My idea on how voxbone works is that voxobone is a intermediatary enabler with the actual hardware with third parties in which case the operator will show as the hardware owner. Is this acuratrate? If you're trying to find the carrier for a number in the UK, start here: http://www.ofcom.org.uk/static/numbering/index.htm Won't cover ported numbers though. S-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
On Fri, 3 Jun 2011, devr devr wrote: thanks for your reply This is the details I get Locality Alloa, Clackmannanshire An intersting place... I spent my youth in a small town near there. (Not that Alloa is a particularly big town to start with!) Central Scotland. Full of history and places with funny names ;-) An odd place to pick, but maybe whoever registered it, just picked one at random in the A's.. (or maybe they live there!) My query now is willl all voxbone numbers show up as the operator as Voxbone SA as above. I wanted to find out who the service provider is on some numbers, I suspected the service to be voxbone but the operator shows as other companies. Then it's probably other companies. There are dozens (100's?) in the UK who can offer VoIP provisioned DIDs - some directly like Voxbone and some via resellers (like me) who resell numbers from the upper-level suppliers... (So if you were to do a lookup on a number that I'd allocated, the 'company' name you'd get back would be me, it would be the wholesaler I'd obtained it from) However what you can't easilly tell right now is who's handling the number if it's been ported - bit of a PITA right now. My idea on how voxbone works is that voxobone is a intermediatary enabler with the actual hardware with third parties in which case the operator will show as the hardware owner. Is this acuratrate? AIUI, Voxbone owns the hardware (and network) that plumbs into the PSTN at one end and provides a VoIP gateway at the other end, and obtained a number allocation from Ofcom. So from a UK point of view, they're at the same level as BT, Virgin, CW, Energis, Magrathea, Gamma, and dozens (100's?) more who've done the same thing. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I use phone line to recive faxes?
Yeah I am using a TDM410P, thanks for the answer. On Fri, Jun 3, 2011 at 4:30 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 02 Jun 2011, khalid touati wrote: Hi Guys, Actually My question is as in the subject, may I use a regular phone line to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8. Yes, you can. BUT, you will need some sort of FXO interface (allows the computer to connect to the telephone socket on the wall), which is supported by DAHDI (or its predecesor, Zaptel). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
On Jun 2, 2011, at 11:24 PM, Satish Barot wrote: With due respect to Digium work, are there no issues with Asterisk 1.8? https://issues.asterisk.org/view_all_bug_page.php And the first of those is a real show stopper at least for us. We've got to have multiple parking lots and that has been broken since the end of last year at least. We opened that ticket on 12/29/10. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue base polycom custom ringtype
Hey Guy, I want to implement Queue base custom ring tone so Agent will get aware of incoming call for sale or tech etc.. I know its possible with SIPAddHeader http://www.technicallyamusing.com/?p=44 I am confused here alertInfo voIpProt.SIP.alertInfo.1.value=custome-ring voIpProt.SIP.alertInfo.1.class=5 We already have alertInfo set to Ring Answer how should i use both ring and Ring Answer ? alertInfo voIpProt.SIP.alertInfo.1.value=Ring Answer voIpProt.SIP.alertInfo.1.class=4/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Thursday, June 02, 2011 11:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 06/02/2011 10:29 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 02, 2011 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. Does Digium run 1.8 on their production corporate PBX? We have two Asterisk systems that comprise our PBX: one is a Switchvox system that handles the bulk of the duties, and there is an Asterisk 1.8 system connected to it that handles all of the stuff Switchvox isn't really designed for. What version of Asterisk does Switchvox use? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
On 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
Sherwood, I was wrong here But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? I have just cross check and we have option OPTIMIZED. That mean don't create coredump right ? -S Date: Fri, 3 Jun 2011 09:53:01 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 Message body On 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
I don't know the statistics involved, but not allowing the compiler to optimize would almost assuredly have some negative effect on performance Sent from my iPhone On Jun 3, 2011, at 10:16 AM, satish patel satish...@hotmail.com wrote: But anyway let me set coredump=yes in asterisk.conf Do you think its a good idea to compile with DON'T OPTIMIZED option in production ? does it impact on performance ? -S CC: asterisk-users@lists.digium.com From: sherwood.mcgo...@gmail.com Date: Fri, 3 Jun 2011 10:13:31 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 No, it just means that the coredump will not have information that is as useful Sent from my iPhone On Jun 3, 2011, at 10:02 AM, satish patel satish...@hotmail.com wrote: Sherwood, I was wrong here But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? I have just cross check and we have option OPTIMIZED. That mean don't create coredump right ? -S Date: Fri, 3 Jun 2011 09:53:01 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
Paul Belanger pabelan...@digium.com writes: Sounds like asterisk was not told to generate a coredump, add the following, then you can generate a backtrace[1]: asterisk.conf [options] dumpcore = yes The challenge with Asterisk and core dumps is that the Asterisk user often does not have permissions to write to the directory it has as current directory. By default, that is where the kernel writes the core dump. You can change the directory by changing the kernel.core_pattern sysctl, but make sure that you pick something which does not present a security threat. It would be very convenient if Asterisk could be told to keep a specific directory as current directory. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about null routing calls to DIDs we don't handle
(reposted with correct subject line, I think messing up the subject
line last time prevented my question from being read. Cheers :)
On Thu, Jun 2, 2011 at 12:27 PM, Jesse Thompson jes...@gmail.com wrote:
Letting a carrier use you as a carrier seems like quite a bad idea
generally..
I think I would agree. :)
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here
get routed upstream
in the 'local' context instead of the other one?
So here is where the finer points of Asterisk pattern matching must
come into play.
All of the customer DID's match the pattern _NXXNXX. If we put
that pattern in the local context, then wouldn't that mean that calls
from a local customer to another local customer would match the
_NXXNXX pattern before even trying to match against the specific
patterns in the clients context? We need to be able to route
local-to-local calls without using two trunks to go back and forth
through the upstream provider.
Thank you for your input. I know this is a problem most operators can
get past, so there's got to be just something not lining up quite
right in my mental model. :)
- - Jesse
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Re: [asterisk-users] Question about null routing calls to DIDs we don't handle
Why not setup a default catch-all route that goes to either your main line (to
drive sales) or a pre-recorded message (the number you dialed is
disconnected...etc), and then setup more specific pattern matches for assigned
numbers? I've done this before for clients that have large blocks of did's
assigned to them but only a small number of extensions that need direct dial
capabilities.
Thanks,
--Warren Selby, dCAP
On Jun 3, 2011, at 2:34 PM, Jesse Thompson jes...@gmail.com wrote:
(reposted with correct subject line, I think messing up the subject
line last time prevented my question from being read. Cheers :)
On Thu, Jun 2, 2011 at 12:27 PM, Jesse Thompson jes...@gmail.com wrote:
Letting a carrier use you as a carrier seems like quite a bad idea
generally..
I think I would agree. :)
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here
get routed upstream
in the 'local' context instead of the other one?
So here is where the finer points of Asterisk pattern matching must
come into play.
All of the customer DID's match the pattern _NXXNXX. If we put
that pattern in the local context, then wouldn't that mean that calls
from a local customer to another local customer would match the
_NXXNXX pattern before even trying to match against the specific
patterns in the clients context? We need to be able to route
local-to-local calls without using two trunks to go back and forth
through the upstream provider.
Thank you for your input. I know this is a problem most operators can
get past, so there's got to be just something not lining up quite
right in my mental model. :)
- - Jesse
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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_
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Re: [asterisk-users] benefits of asterisk 1.8
On Fri, 2011-06-03 at 09:07 +0100, Ishfaq Malik wrote: Are you suggesting that there are no bugs in 1.4 or 1.6? I presume that you are aware of the fact that it is impossible to prove the absence of bugs in any piece of software You might not have detected them yet. Furthermore behaviour that might have been coded on purpose, can be considered eroneously some time later. Currently there seems to be a fear of 1.8. We're about to put it into production and yes, we've had issues with it, mostly due to the fact we use RealTime, but before you change anything it is always advisable to test the hell out of it. To anyone who is thinking of moving to 1.8 the question is not, 'is it stable?'. The question is, 'have I comprehensively tested it to show that it is suitable for my needs?' If you put it into production, test at least the functions that you are going to use. There might (and probably will) problems in the code, but as long as it does not bother you, so what? And don't stop testing after you put it into production: have a shadow system (with representative configuration). According to Murphy, side-effects will probably rise to the survice after going into production End-users will come up with situations you never enticipated in your worst nightmares. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
