Re: [Asterisk-Users] Queues and timeouts
Adam Goryachev wrote: [snip] This busy means, tell the queue app that the agent is busy. The queue app willl go try someone else. The caller will keep hearing music. :) Julian, and others, If someone offers you a suggestion towards solving your problem, you might at least try it before discarding the advice. I (or the person replying to you) went to some effort to learn this stuff at some point, and to type the reply to you. I must turn round and defend myself here against this - I normally do listen to all the advice I'm given, but this simply did not make sense to me at the time. It's that simple. I was also not in a position to try out said advice because I was a hundred miles away from the machine with no ssh access - the wife forced me to take a holiday .. ;) [ok, so I deleted the rest of my rant...] perhaps just as well. ;) Please, let's just get along, and try things out, even if we thing it won't work... I do investigate problems. In fact, although I'm not a "C" coder I have tracked down *and fixed* two bugs in the Agent / Q systems and have had the fixes comitted to CVS. That doesn't happen by accident. PS, ok, if I suggest that rm -rf / will solve all your problems, then maybe you shouldn't try it :) Works for me >-) Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and timeouts
PLEASE FOR THE LOVE OF GOD put a NAME in your email program.. I'm sure it makes going back and finding stuff in the archives when you and about 100 other people use "Asterisk" in their names This goes for anyone that uses "Asterisk", "Asterisk PBX" or any form there of .. lets put a name in there. /b On Jul 24, 2005, at 1:44 AM, Asterisk wrote: Joseph wrote: [snip] exten => _6XXX,2,Busy exten => _6XXX,3,Hangup But the whole point is that I don't want the caller to hear a busy signal or get hung up, I want the Queue to try the next available agent. Which it does at the moment, just with the errors mentioned in the error log file. This busy means, tell the queue app that the agent is busy. The queue app willl go try someone else. The caller will keep hearing music. :) damn, that's so obvious when you say it - I'm sorry that I questioned you, but it smelt wrong ;) Many thanks. I'll go try that now. Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and timeouts
Joseph wrote: [snip] exten => _6XXX,2,Busy exten => _6XXX,3,Hangup But the whole point is that I don't want the caller to hear a busy signal or get hung up, I want the Queue to try the next available agent. Which it does at the moment, just with the errors mentioned in the error log file. This busy means, tell the queue app that the agent is busy. The queue app willl go try someone else. The caller will keep hearing music. :) damn, that's so obvious when you say it - I'm sorry that I questioned you, but it smelt wrong ;) Many thanks. I'll go try that now. Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ClueCon in 2 Weeks!
I'll talk to your boss if he has a problem! ;) /b On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote: Mine did. [EMAIL PROTECTED] 7/21/2005 2:54 PM >>> Brian West wrote: ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! Thanks, Brian West Asterlink.com Anyone else think that was a joke at first impression? Good luck convincing the boss to pay for your way to "ClueCon" ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Lukins & Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] XML or Push Info
Guys. Which phones support pushing information or xml push info? For making info applications and such, of course, besides Cisco? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: 2 asterisks connected but trying to bridge
Also, both asterisks have notransfer?yes and I get this -- Attempting native bridge of IAX2/[EMAIL PROTECTED] and IAX2/voipjet-9 Why? Seems its not taking the notransfer into account. Now Im puzzled |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Vincent Luba |Sent: Viernes, 15 de Julio de 2005 03:19 a.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] RE: 2 asterisks connected but trying |to bridge | |Hey, |For the bridge issue, take a look at 'notransfer=yes' option |in your iax.conf. |It'll force * to stay in the path |http://www.mail-archive.com/asterisk-users@lists.digium.com/msg 42262.html | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX phone not hear the other phone ring when calling
option r. 'nuff said. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+dial On Sat, 23 Jul 2005 14:48:10 -0500, "Maps" <[EMAIL PROTECTED]> said: > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension Lights Patch
Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ClueCon in 2 Weeks!
Mine did. >>> [EMAIL PROTECTED] 7/21/2005 2:54 PM >>> Brian West wrote: > ClueCon is coming in 2 weeks so we urge everyone who plans on > attending to register today so we get a proper headcount! > > > > Thanks, > Brian West > Asterlink.com > Anyone else think that was a joke at first impression? Good luck convincing the boss to pay for your way to "ClueCon" ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Lukins & Annis, P.S. NOTICE: This email may contain confidential or privileged material, and is intended solely for use by the above referenced recipient. Any review, copying, printing, disclosure, distri- bution, or any other use, is strictly prohibited. If you are not the recipient, and believe that you have received this in error, please notify the sender and delete the copy you received. Thank You! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dead pg connection with voicemail
Title: Message When using Postgres as the backend to voicemail.conf using [general] dboption=dbname=asteriskdb user=asterisk password=asterisk Everything works great until I restart Postgres. In this case asterisk does not attempt to reconnect to the database even when it knows that there is a dead connection. I receive the following message at the console. PGSQL_query: Connection Error (no connection to the server) And I can not get it to reset even with a "reload" command on the CLI. I have to restart asterisk. Has anyone found a way to work around this "feature" that only connects to the postgresql database at asterisk startup for voicemail? Thanks Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callgen323 & ohphone!!
Hello, I just wanted to know if anybody is using callgen323 and ohphone. I am running callgen323 on 2 machines with gatekeeper on one of the machines. When callgen is started, it registers with gatekeeper and places multiple call sessions. Ohphone has capability of placing single call and shows, RTP audio sesssion statistics like packet sent, packet received, jitter etc. Can callgen323 show those statistics? If so, could anybody let me know how? Also, does anybody have clue of "ogm.wav" file in callgen323? Can you let me know where to find to it. I would really appreciate you help in this matter. Thanks Radhika ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Business Edition
"Brian West" <[EMAIL PROTECTED]>: Aidan isn't a troll he does raise a very valid point. Which was, I presume, that companies that once collaborated on Asterisk development such as Sangoma don't find themselves on friendly terms with Digium now that they're competing for * implementors $$s? If Aidan's point was that Digium isn't playing nice nice with Sangoma, I think that's politics that most of us don't know enough to comment on, and that should not really concern anyone but the companies involved. So maybe he _is_ a troll. If his point was that the dual licensing model somehow disadvantages companies such as Sangoma, who don't find themselves on a level playing field, and that this causes pain and suffering to * users because of lack of competition in hardware supporting asterisk, well them maybe he's not a troll. Please don't get all protectionist on me and say Digium has earned the right to be the only hardware vendor. I'll respect your right to "Buy Digium" as long as you respect my right to buy the cheapest or perhaps the most critically acclaimed card asterisk-supporting flavor of the month ;-) I know I won't hurt Mark's feelings - he recognizes that competition is a healthy part of the capitalist market Digium is in the process of disrupting. The following is excerpted from: http://news.com.com/Is+the+telephone+industry+ready+for+open+source/2008-1082_3-5737703.html - "Capitalism, to me, is a system of competition in which the end product becomes better and (less expensive) because of competitive forces pushing on it. Open source forces you to have to be more competitive. As you add competitive advantage, that advantage is adopted more rapidly; you can't sit back on your development. It forces you to constantly be innovative because people can use your new thing immediately." -- I think we've already seen the benefits of this competition - we've seen Digium innovate their hardware quite rapidly, adding the often-requested echo cancellation to their already popular T1/E1/J1 boards for instance. But I guess I'm wondering ... does the present licensing model discourage other vendors from contributing to *? I'm not sure Sangoma developers could sign the disclaimers even if they wanted to ... but then again I don't know if there's anyone there with anything to offer. I would think that that fact that they're selling hardware that supports * means that there's _some_ sharp cookies there, but perhaps they're just kernel module/driver hackers out to make a quick buck off of Digiums's back without contributing to the core? -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
Or better yet.. modify the disclaimer like I and a few others did to say that the only thing you will disclaim are things you post on the bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted under your user on mantis IT IS NOT DISCLAIMED! /b On Jul 23, 2005, at 2:59 PM, William Lloyd wrote: On 23-Jul-05, at 11:22 AM, Kevin Walsh wrote: On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? You're forgetting about the "disclaimer" documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform "the owner", and would secondly have a prior agreement with "the owner" to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. It's unlikely that the current pool of asterisk developers will remain static however. People change jobs, new people find asterisk interesting, people that have not contributed before start to contribute. Assuming a fork were to happen one day. Lots of current developers would stay with the Digium tree because they know it, are digium partners, think it's a better idea, already signed the disclaimer and don;t have an issue with it etc. Many new developers submitting smaller patches would not bother to sign a legal disclaimer and just submit the patch to the full GPL tree. The splinter GPL tree would likely integrate the changes faster and obviously don;t care about a disclaimer. The practicalities of tracking the changes between two source trees would just get more and more time consuming for Digium. They will want to make 100% legal sure that every change they bring into their tree comes from somebody with a disclaimer. Rewriting the missing bits with other programmers would just help the tree's diverge faster. Meanwhile a full GPL tree can just plow ahead without concern. Many companies successfully manage the commercial GPL gap. MySQL for example. The difference in this case is selling a binary only version instead of making money off just hardware and support services/contracts. At the end of the day Digium own the Asterisk trademark and in the world these days, brand name recognition is often more important than the product behind it. -bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Business Edition
Aidan isn't a troll he does raise a very valid point. /b On Jul 23, 2005, at 5:55 PM, Brian Capouch wrote: Aidan Van Dyk wrote: Is this indicative to how Digium people respond to everything (including the company that built the first asterisk-supporting hardware still continuing to make hardware which Asterisk works on)? Nothwithstanding the almost-unparseable syntax, let's not feed this troll . . . b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opteron Hardware with Asterisk
Sarge. RHEL/compatible should be fine too. Wiley Siler wrote: Did you build it using the 64 bit CentOS or another Distro? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Dobrin Sent: Friday, July 22, 2005 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Opteron Hardware with Asterisk I have asterisk running on dual 244's. Everything works fine, the only special issue i had was installing the g729a codec (required a very tiny tweak to the asterisk Maiefile). Unfortunately, the system doesn't get a huge amount of traffic, so I can't testify to capacity. Running 1.0.8, btw. Asterisk Supporter wrote: Anyone running Asterisk on dual Opteron Server? Are there any special issues in a 64 bit environment and what is the capacity curve like? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS-HEAD dies signal 11 after incorrect vm password
Hello, On Fri, 22 Jul 2005 19:49:15 -0400 Adam Dobrin <[EMAIL PROTECTED]> wrote: | anyone else have the above issue? this is today's CVS. | | thanks. Well, i found that as well couple days ago, and even reported that on #asterisk-dev. A full backtrace can be found here: http://tinyurl.com/7o2k7 Hope that helps. Thanks, Philippe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Opteron Hardware with Asterisk
Did you build it using the 64 bit CentOS or another Distro? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Dobrin Sent: Friday, July 22, 2005 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Opteron Hardware with Asterisk I have asterisk running on dual 244's. Everything works fine, the only special issue i had was installing the g729a codec (required a very tiny tweak to the asterisk Maiefile). Unfortunately, the system doesn't get a huge amount of traffic, so I can't testify to capacity. Running 1.0.8, btw. Asterisk Supporter wrote: >Anyone running Asterisk on dual Opteron Server? Are there any special >issues in a 64 bit environment and what is the capacity curve like? >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc:sync lost, pci performance too low. you might have some cpu throtteling enabled.
Hi Terry, thanks for the info - might - so shall get an Intel board/cpu combination or a quadbri card - or might it be the case, that the 2400+ athlon xp is running only as 1800+ cause I didn't update the mainboard bios? yours, Alex Terry Wade wrote: Spoke to Klaus-Peter about this PCI performance issue. He says it has to do with the CPU not supporting cpufreq stepping. I had to get a quad card to get the issue resolved. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Szlezak Sent: 17 July 2005 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc:sync lost,pci performance too low. you might have some cpu throtteling enabled. HI Hartmut, I do have the same problem as you decribed earlier. The Billion HFC Cards (two of them) work flawlessly in my old Pentium II, but in my more powerful Athlon XP 2400+ (Via KT400 Chipset), I allways get the "pci performance too low" message and syslog kills the system. I'll try it now with commenting out the message. Do you know anything more in the meantime. Did Mr. Junghanns have any statement about it? I wonder if chan_mISDN is a better choice by now? Thanks for any advice you might have! yours, Alexander Hartmut Wahl wrote: Hello, I have investigated the issue a bit further, I was not able to find the root cause, maybe it is the KT133 Chipset of my ASUS A7V. However I found a bad hack to make it work under some circumstances. I commented out the line: printk(KERN_CRIT "zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.\n"); in zaphfc/zaphfc.c. Since when this situation happened once, the syslog started and this caused the situation again -> endless loop with high sysload. Now the sync lost happens probably every now and then but it does not go into an endless loop. I also recognized that I must not run setiathome since then the audio quality of connections via the hfc-card will suffer (crackling). Amazingly cpuburn (takes every cpu-Time it gets as well) does not have this effect. I have no idea what strange things setiathome does to cause this but it reminds me of a sound card problem on this board. My SB-Live had crackling when I ran setiathome but I think with a newer driver (and ALSA I think) this problem was gone. Things that did not help: - Trying to change the latency (is fixed to 16 and cannot be changed) - Trying to change the latency of my other pci-devices (much higher and much lower). - Playing around with BIOS Options like delayed transaction, etc. - Changing PCI-slots - Making sure that the ISDN-card did not share the IRQ - Having only the ISDN-card and the Video-card in the System Hope that helps others who encounter this problem as well Hartmut On Sat, Sep 11, 2004 at 12:55:21PM +0200, Hartmut Wahl wrote: Hi! On Sat, Sep 11, 2004 at 10:38:48AM +0200, Maurizio Marini wrote: my machine did hangup as growing logs fullfilled partition hmm I see, mine is 8G, but it has gronwn from 0.5G to 2.0G it does apply to asterisk, not to zaphfc :( it was a misleading suggestion, so i solved it installing in an other more powerful machine: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 8 model name : Pentium III (Coppermine) stepping: 10 cpu MHz : 999.556 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse bogomips: 1957.88 with this hw i've no issues at all; even strange messages i complained about in my previous posts like: ok we are getting closer, although speed shouldn't be a problem: vendor_id : AuthenticAMD model name : AMD Athlon(tm) Processor stepping: 4 cpu MHz : 1059.618 cache size : 256 KB bogomips: 2097.15 it is a 1,4GHz underclocked, since my ASUS A7V with KT133 does only support 100MHz FSB, but it requires only slow and quite fans. I think I'll try different kernels maybe something in the Debian k7-kernel-image interferes. Greetings Hartmut ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.di
Re: [Asterisk-Users] Re: Re: Business Edition
Aidan Van Dyk wrote: Is this indicative to how Digium people respond to everything (including the company that built the first asterisk-supporting hardware still continuing to make hardware which Asterisk works on)? Nothwithstanding the almost-unparseable syntax, let's not feed this troll . . . b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Business Edition
Kevin P. Fleming wrote: > Aidan Van Dyk wrote: >> I guess I read 1 more FAQ than you. But I'm not sure how quoting a >> Google FAQ is libelous FUD. > > Your original message did not quote a Google FAQ, it accused Digium of > joining the SoC program only to collect the $500 mentoring fees. I > believe that qualifies :-) My original message on the thread: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user:110284 where I quoted that FAQ, and specifically asked if Digium had worked out anything with Google regarding this. And when I ask a followup question to the previously answered one, it's taken as libel? Is this indicative to how Digium people respond to everything (including the company that built the first asterisk-supporting hardware still continuing to make hardware which Asterisk works on)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC gives me only the time, but no cost
Thank you very, very much Rusty. I reopened the bug report.
http://bugs.digium.com/view.php?id=4479 I made a very slight change to
the method it uses to calculate costs but it should implement the
connect charge properly. Initially I rewrote the cost calculation code
but that was not necessary, it can be implemented by changing the
following lines
my $adjtime = int(($answeredtime + $increment - 1) / $increment) *
$increment
becomes
$adjtime = int((($answeredtime - $numdata->{includedseconds}) +
$increment - 1) / $increment) * $increment
Anybody that is interested, please feel free to comment on the bug
report. Once that one goes in, another bug will probably get commited
that allows you to calculate what ASTCC will charge for a call from the
web interface. That makes testing and confirming that you are setting
it up right much easier.
Darren Wiebe
[EMAIL PROTECTED]
Rusty Shackleford wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wiebe
Sent: Saturday, July 23, 2005 8:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ASTCC gives me only the time,
but no cost
The included seconds field is not taken into account when billing the
connect charge. IMHO this is a bug but I've not gotten
enough feedback
to put the patch through. Therefore the patch has been closed. :-)
I spent an afternoon going through that code again, Darren. You were
right. If we assume that the intent was to use the "includedseconds"
column value as a way to allow for x/y billing intervals, and set
"connectcost" to the value that we want to charge for the call minimum
charge - "x", the stock code charges that amount, but also starts the
meter running on the "y" value from the start of the call, resulting in
an over charge. The "y" value, by the way, is set in the "brands" table
and flows to the "cards" table when cards are created (breaking
normalization). For example, we have a route to McMurdo Station for
which we charge $.50 per minute, in six second increments with a 30
second minimum (30/6). If we set the connect charge column to $.25, the
included seconds to 30, and the cost to $.50, a 30 second call should
cost $.25. Instead, it's costing $.50, because ASTCC charged the connect
fee, plus the cost of 5 six second increments - $.25. It shouldn't start
charging those six second increments until AFTER the "includedseconds"
interval has passed.
I've patched by scripts to correct this. It would be nice if the
correction were made to the distributed source, perhaps with some
documentation of how things are supposed to work.
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Re: [Asterisk-Users] IAX over HTTP
Dave Cotton wrote: On Fri, 2005-07-22 at 15:42 -0500, Eric Wieling aka ManxPower wrote: Eric Rees wrote: We have been running IAX through OpenVPN with SSL for 6 months without any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX sounds better then the land line. Using UDP or using TCP? Might want to confirm by using tcpdump. OpenVPN uses UDP OpenVPN can use TCP, and really, I would expect that many users using openvpn to bypass firewall rules, would be using TCP not UDP. Simple example... - dev tun proto tcp-server port 443 - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
On 23-Jul-05, at 11:22 AM, Kevin Walsh wrote: On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: Adam Goryachev [EMAIL PROTECTED] wrote: On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: For this reason, I believe that if a fork were ever necessary, it would struggle to beat a distinct path away from the Asterisk Binary Edition Correct, until the point where there is MORE features being added to the forked version of asterisk than the digium version of asterisk. That can't happen, because the ABE could, and probably would, absorb all of the advances in the fork, while forging ahead with the original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? You're forgetting about the "disclaimer" documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform "the owner", and would secondly have a prior agreement with "the owner" to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. It's unlikely that the current pool of asterisk developers will remain static however. People change jobs, new people find asterisk interesting, people that have not contributed before start to contribute. Assuming a fork were to happen one day. Lots of current developers would stay with the Digium tree because they know it, are digium partners, think it's a better idea, already signed the disclaimer and don;t have an issue with it etc. Many new developers submitting smaller patches would not bother to sign a legal disclaimer and just submit the patch to the full GPL tree. The splinter GPL tree would likely integrate the changes faster and obviously don;t care about a disclaimer. The practicalities of tracking the changes between two source trees would just get more and more time consuming for Digium. They will want to make 100% legal sure that every change they bring into their tree comes from somebody with a disclaimer. Rewriting the missing bits with other programmers would just help the tree's diverge faster. Meanwhile a full GPL tree can just plow ahead without concern. Many companies successfully manage the commercial GPL gap. MySQL for example. The difference in this case is selling a binary only version instead of making money off just hardware and support services/contracts. At the end of the day Digium own the Asterisk trademark and in the world these days, brand name recognition is often more important than the product behind it. -bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX phone not hear the other phone ring when calling
Dear Helpers! I have setup my asterisk with the iax phone. However, when I try to dial out to the pstn phone from the iax phone, I can't hear the other phone ring until they pickup the phone. Doesn't anyone know the issue? Doesn't anyone has experience with the problem? Please helping me or why and how I could correct the problem? I would very appreciated! Regards, Lan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
Try Florz patch with your bristuffed asterisk. Better support for missed interrupts. Julian J. M. On 7/22/05, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: > Hi, > I tried to install a tdm400P and a monoBRI. I loaded zaphfc and wcfxs > modules and everything seemed allright but linux log shows the following > message: > zaphfc: sync lost, pci performance too low. you might have some cpu > throtteling enabled. > Anybody knows what it means? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
The system has been rock solid on both performance and reliability. One is located in out corporate office, two in local data center, and three distributed California, Wyoming, and NYC. We have experienced issues with some of our ISPs but nothing that could blamed on the hardware. I will likely continue to order Dell as we expand the system. Mainly for uniformity though. Email me off list if you like to see a test. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Friday, July 22, 2005 4:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell Hardware No audio latency? No drops? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Michael Miller |Sent: Viernes, 22 de Julio de 2005 02:33 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Dell Hardware | |We have 6 dual proc Dell 1850s with a TE410P in each and they |have worked without fault. I know that Digium has a |compatibility note on the web site regarding the NIC but I |have not seen any issues. Our largest conference with a |mixture of Zap, SIP, and IAX clients was close to 200 |participants on a single server had no issues. | |Michael | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Friday, July 22, 2005 9:48 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Dell Hardware | |Guys. | |What do you think about Dell hardware and Asterisk? Whos using |it, comments, any special specs recommended or models? | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about the latest CVS and Zaptel
I'm having trouble with the latest cvs HEAD (7/22/05) and my Wildcard TE405P I just got in from Digium. I'm not able to get podprobe to work with the release. I get an error "unable to install" however when I grab the stable it works great but no realtime drivers for asterisk. I also tried to just get the stable of zaptel and the HEAD of asterisk but asterisk would not load. Any one have any tips? Thanks for taking the time to read this message! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 Server Re: Dell Hardware
M O wrote: Hello, Just chiming in here: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Viernes, 22 de Julio de 2005 01:24 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dell Hardware Mmhh nice !! So, why did Digium forbid it :)? If Dell is so bad... why is a Dell 2850 server one of the two listed on the compatibility list for ABE? http://www.digium.com/index.php?menu=product_detail&category=software&produc t=ABE&tab=compatibility I am running Asterisk on a 100Mbps Pipe on the following: Hardware Information Processors 4 <-(should be 2) Model Intel(R) Xeon(TM) CPU 3.00GHz Chip MHz2992.81 MHz Cache Size 1024 KB System Bogomips 23907.52 PCI Devices 00:1f.1 IDE interface: Intel Corp. 82801EB/ER 02:0e.0 RAID bus controller: Dell PowerEdge Expandable RAID controller 4 06:07.0 Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller 07:08.0 Ethernet controller: Intel Corp. 82541GI/PI Gigabit Ethernet Controller 0b:0d.0 VGA compatible controller: ATI Technologies Inc Radeon RV100 QY [Radeon 7000/VE] IDE Devices hda: TEAC CD-ROM CD-224E SCSI DevicesMegaRAID LD 0 RAID0 69G (Direct-Access) PE/PV 1x6 SCSI BP (Processor) USB Devices Linux 2.4.21-27.0.1.ELsmp ehci-hcd Intel Corp. 82801EB USB2 00:1d.7 USB UHCI Root Hub bca0 USB UHCI Root Hub bcc0 Sincerely, Martin I'm doing just fine on the exact same hardware with RHEL 4 and CVS-HEAD. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog extensions behind E1, how to create them?
I will have some extensions behind an E1. All of them will need the features/applications of Asterisk. Analog Extensions <-> PABX E1 <-> E1 Asterisk IP <-> VoIP trunk ^ | | IP Phones How is the best way to create this users on Asterisk? Some of them will have a SIP account to have its extensions with mobile functionality when they will be out of office, others will not have this feature. Some examples will be great! Thanks. Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing SIP Problems with Asterisk and SER on same PC
[EMAIL PROTECTED] wrote:
> Hello fellow asterisk people!
>
> I have Asterisk listening on port 5061 and SER on port 5060.
>
> Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
>
> My problems are with SIP. I can make incoming calls from SIP to asterisk
> and to any of the other networks, but when I try to make an outgoing
> call from Asterisk to SER I see the following in Asterisk:
>
> -- Executing Ringing("H323/ip$192.219.85.57:2712/8570", "") in new
> stack
> -- Executing Dial("H323/ip$192.219.85.57:2712/8570",
> "sip/[EMAIL PROTECTED]|20|r") in new stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 482 "Loop Detected" back from 192.219.85.57
> -- Now forwarding H323/ip$192.219.85.57:2712/8570 to
> 'Local/[EMAIL PROTECTED]' (thanks to SIP/sip_proxy-out-f67d)
> Jul 22 20:20:25 NOTICE[29756]: chan_local.c:378 local_alloc: No such
> extension/context [EMAIL PROTECTED] creating local channel
>
> Jul 22 20:20:25 NOTICE[29756]: app_dial.c:232 wait_for_answer: Unable to
> create local channel for call forward to 'Local/[EMAIL PROTECTED]'
>
Read what Asterisk is trying to tell you! There's no extension 280 in
context sip-incoming for the local channel... Fix that.
If you after that still have SIP problems, please try to add SIP debug
output, otherwise it's usually very hard to help.
/Olle
Astricon 2005 - where the Asterisk community meets!
http://www.astricon.net/2005/
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[Asterisk-Users] Outgoing SIP Problems with Asterisk and SER on same PC
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other networks, but when I try to make an outgoing
call from Asterisk to SER I see the following in Asterisk:
-- Executing Ringing("H323/ip$192.219.85.57:2712/8570", "") in new
stack
-- Executing Dial("H323/ip$192.219.85.57:2712/8570",
"sip/[EMAIL PROTECTED]|20|r") in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 482 "Loop Detected" back from 192.219.85.57
-- Now forwarding H323/ip$192.219.85.57:2712/8570 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/sip_proxy-out-f67d)
Jul 22 20:20:25 NOTICE[29756]: chan_local.c:378 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
Jul 22 20:20:25 NOTICE[29756]: app_dial.c:232 wait_for_answer: Unable to
create local channel for call forward to 'Local/[EMAIL PROTECTED]'
== Everyone is busy/congested at this time
-- Timeout on H323/ip$192.219.85.57:2712/8570
== CDR updated on H323/ip$192.219.85.57:2712/8570
-- Executing Goto("H323/ip$192.219.85.57:2712/8570", "#|1") in new
stack -- Goto (default,#,1)
-- Executing Playback("H323/ip$192.219.85.57:2712/8570",
"demo-thanks") in new stack
-- Playing 'demo-thanks' (language 'en')
== Spawn extension (default, #, 1) exited non-zero on
'H323/ip$192.219.85.57:2712/8570'
Want I want to happen is the call to go out through Asterisk - to SER
(as SER knows where the SIP extension is) - and then onto the extension
of the person to call.
In my sip.conf I have the following:
[general]
context=sip-incoming; Default context for incoming
calls autocreatepeer=yes
recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;realm=fedcore2.eicon.com ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain
name
port=5061 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
disallow=all; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
register =>Asterisk:[EMAIL PROTECTED]/5000
[sip_proxy-out]
type=friend ; we only want to call out, not be
called secret=
username=Asterisk ; Authentication user for outbound
proxies host=fedcore2.XXX.com
In my extensions.conf I have
exten =>_5XXX,2,Dial(sip/${EXTEN:[EMAIL PROTECTED],20,r)
So that dialing an extension 5XXX rings sip extension XXX.
I also the following context to catch incoming SIP calls.
[sip-incoming]
exten=>s,1,Wait,1
exten =>s,2,Goto(default,384220,1)
exten =>5000,1,Goto(default,384220,1)
exten =>_9.,1,Goto(default,${EXTEN:1},1)
Why am I unable to make outgoing SIP calls?
I have also not made any changes to my DNS SVR settings (in case I need
to???)
I have also tried to have the following alternative in my extensions.conf
It is something like this:
Asterisk extensions.conf:
[globals]
SERADDRESS=XXX.XXX.XXX.XXX:5060
[context]
exten =>_5XXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,r)
However, this then causes a message "Message to Big" to be displayed in SER.
Many thanks for your help. I am probably doing something obvious wrong!
Thanks
David
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RE: [Asterisk-Users] ASTCC gives me only the time, but no cost
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Darren Wiebe > Sent: Saturday, July 23, 2005 8:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] ASTCC gives me only the time, > but no cost > > > The included seconds field is not taken into account when billing the > connect charge. IMHO this is a bug but I've not gotten > enough feedback > to put the patch through. Therefore the patch has been closed. :-) I spent an afternoon going through that code again, Darren. You were right. If we assume that the intent was to use the "includedseconds" column value as a way to allow for x/y billing intervals, and set "connectcost" to the value that we want to charge for the call minimum charge - "x", the stock code charges that amount, but also starts the meter running on the "y" value from the start of the call, resulting in an over charge. The "y" value, by the way, is set in the "brands" table and flows to the "cards" table when cards are created (breaking normalization). For example, we have a route to McMurdo Station for which we charge $.50 per minute, in six second increments with a 30 second minimum (30/6). If we set the connect charge column to $.25, the included seconds to 30, and the cost to $.50, a 30 second call should cost $.25. Instead, it's costing $.50, because ASTCC charged the connect fee, plus the cost of 5 six second increments - $.25. It shouldn't start charging those six second increments until AFTER the "includedseconds" interval has passed. I've patched by scripts to correct this. It would be nice if the correction were made to the distributed source, perhaps with some documentation of how things are supposed to work. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 is getting closer - please help
Olle, Awesome! Now that everybody know your aiming for September 1 for Asterisk 1.2, I'm sure will make it. Come' on Asterisk community, step up to the plate! PB Olle E. Johansson wrote: > Dear Asterisk Community, > > Asterisk 1.0 was released at Astricon 2004, in September last year. It's > been almost a year and we haven't been able to go ahead and release a > new version. Now is the time to try to move forward again. > > As we've outlined before, the process is this: > > * Code freeze: At this point, we'll stop accepting new additions > (new functions) to the source code. Bug fixes are more than > welcome, but additions will be postponed until after release > and added to the 1.3dev source code base (the new HEAD). > > * Release candidate: A release candidate will be produced as a > tar.gz file on the FTP site. > > * Release of 1.2: The new release version of Asterisk, that replaces > Asterisk 1.0 > > * Release of 1.2.1: The working version :-) of the new version of > Asterisk > > > > So why 1.2.1? Well, the common feeling among developers is that > "No one really tests anything until we release, so we will receive > bug reports from the hour we release 1.2.0". Let's try to prove > that they are wrong! > > What can you do to help this process? > - > * Set up a test system, and test CVS head in something that resembles > your production environment. Scripts, phone, dialplan - make sure > you use as many of the features as you can and use in production > to make sure they work as expected in version 1.2 > > * Go wild and test at least two of the new features in 1.2 just > for fun and make sure they work as documented. Or document how > they work if it's not documented. Test the new realtime architecture, > voicemail ODBC storage, AEL - the new scripting language, > the new dialplan templates and constructs, the #exec config directive, > attended transfers, native music on hold... The list is long. > > * If you have reported bugs or filed patches in the bugtracker > (bugs.digium.com), make sure you reply quickly when a bug marshal > or developer ask you questions or require more information. > At this point, we're working very hard to clear out outstanding > bugs and stabilize the additions that is waiting for inclusion > in the CVS. We will close reports that we can't move forward if > we do not get any responses. We can re-open later, but need to > move forward. If we have a report of a proven bug that needs fixing, > those will not be closed. Only unclear reports with no responses > will be closed. > > * Visit the bug tracker at bugs.digium.com and help us test patches. > Postitive and negative reports are both equally needed. There's no > way a small team of core developers and bug marshals can test > everything in there now. We need to decide which patches that are > ready for inclusion, that are tested and documented. > > * If you find that we're missing documentation, please add to the > readme files, write new ones. The Asterisk documentation team is > ready to help you if you need assistance in this effort. > > * Disappoint the developers by making sure that the CVS head gets > a thorough testing phase now, before release! > > * Update the Wiki on the 1.2 version. Make sure that you make it > very clear that new features only work in 1.2 and releases after > that so you won't confuse readers that use older versions. > > * Test Asterisk CVS head on other platforms than Linux: FreeBSD, > OpenBSD, MacOS/X, Commodore VIC 20 - will it work? > > When is 1.2 scheduled to be released? > - > At usual with Open Source, we release when the software is ready for > release. We do not release when it suits the marketing department, when > we need a positive stock report or when customers require it. > > That said, we now are trying to focus on getting a release out of the > door around September 1st. No promises, it all depends on your help and > assistance to move forward. Please ask your boss for some time and > resources to help the project with testing or dedicate resources within > your company to help us. It's Open Source, meaning that everyone works > together to make sure we get the software that works for our home, our > company or our organization. > > Finding information > --- > If you have questions about the developer version, the base for the > 1.2 release, use the #asterisk-dev channel on the freenode.net IRC. If > you have questions about bug reports and patches, find a bug marshal > in the #asterisk-bugs channel. To find out how to download or connect > to the IRC channel, please visit http://www.asterisk.org > > Thank you for your assistance! > > /Olle > > - > Astricon 2005 - W
Re: [Asterisk-Users] astcc timestamps
I'm not sure on sorting it but it is fairly easy to change in the source
code. Here is what you would have to do to change the format.
1. replace this line in astcc.agi:
$callstart = localtime();
with:
$callstart = timestamp();
2. Add this subroutine to astcc.agi:
sub timestamp() {
my $now = strftime "%Y%m%d%H%M%S", gmtime;
return $now;
}
3. Add this command near the beginning of astcc.agi with the other
"use" commands.
use POSIX qw(strftime);
Should change it to the format you are looking for.
Darren Wiebe
[EMAIL PROTECTED]
Ronald Wiplinger wrote:
The time stamps in ASTCC are useless as they are now:
Fri Jul 22 15:06:25 2005
Wouldn't it be better to use something like:
2005-07-22 15:06:24 Fri
I want to sort the records by date, but with the format now it is
impossible... or do I miss something?
bye
Ronald Wiplinger
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Re: [Asterisk-Users] X100P not answering
On Sat, 23 Jul 2005 13:21:56 +0300, Tzafrir Cohen wrote
> On Fri, Jul 22, 2005 at 05:39:43PM -0500, Carlos Chavez wrote:
> > I have an Asterisk server running todays CVS (updated it just in case
> > that was the problem). It has 3 X100P cards. The first two lines I use
> > as my normal work lines and the third is my fax line which I use with
> > SpanDSP. I run Fedora Core 4.
>
> Interested to know how this works. There seem to be quite a few problems
> for spandsp through a TDM card, right?
>
I have used SpanDSP for the last 5 months with about 98% success at
receiving faxes. To avoid sqashed TIFF files I convert them to PDF before
emailing them. I do not use SpanDSP for sending, I prefer efax for that.
> >
> > I have a problem that the third X100P does not answer the call. From
> > the console I can see that there is an incoming call with the following:
> >
> > -- Starting simple switch on 'Zap/3-1'
> > Jul 22 17:34:21 ERROR[30490]: callerid.c:266 callerid_feed: fsk_serie
> > made mylen < 0 (-16)
> > Jul 22 17:34:21 WARNING[30490]: chan_zap.c:5827 ss_thread: CallerID feed
> > failed: Success
> > Jul 22 17:34:21 WARNING[30490]: chan_zap.c:5871 ss_thread: CallerID
> > returned with error on channel 'Zap/3-1'
>
> Any chance that this is the same issue as the UK caller ID issue? I seem
> to recall that the same problem applies to Mexican phone lines as well.
No, actually my fax line does not have Caller ID activated. CallerID
works like a charm in Mexico as we use basically the same system as the USA.
>
> > -- Executing Wait("Zap/3-1", "1") in new stack
> > -- Executing Answer("Zap/3-1", "") in new stack
> > -- Executing ResponseTimeout("Zap/3-1", "5") in new stack
> > -- Set Response Timeout to 5
> > -- Executing SetLanguage("Zap/3-1", "es") in new stack
> > -- Executing AbsoluteTimeout("Zap/3-1", "600") in new stack
> > -- Set Absolute Timeout to 600
> > -- Executing BackGround("Zap/3-1", "bien-fax") in new stack
> > -- Playing 'bien-fax' (language 'es')
> > Jul 22 17:34:25 WARNING[30490]: chan_zap.c:3696 zt_handle_event:
> > Ring/Off-hook in strange state 6 on channel 3
> > == Spawn extension (fax, s, 6) exited non-zero on 'Zap/3-1'
> > -- Hungup 'Zap/3-1'
> >
> > The calling phone just keeps ringing all the time and the messages
> > above are repeated until I hangup the call. My other two lines work
> > perfectly. Any ideas?
>
> Need an Answer in the dialplan?
>
The second comand after Wait is Answer. I do not really need the wait as
this line does not have caller ID, but I left it there anyway.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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[Asterisk-Users] Need to start from somewhere
Hello all, since we are sitated in Serbia and Montenegro and Linux, VoIP, Open Source and ECONOMY here are in the diapers. We are traying to start "Asterisk implementation project" and make it available to small people (in small companies,hotels,homes etc). The reason is that here we have "exclusive distributors" of PBX equipment (Panasonic, Ericsson,Cisco,3com etc.) in pocket and cicuite comutation, and for them sky is the limit when the price of equipment is in question , they have money for orders and sell equipment at much,much higher price if you compare it with EU prices or USA prices. And people like we have needs and do not have that much money, so we are traing to make alternative and hope it will be better ... SORRY FOR MY LONG PRELUDE ( we respect kbps) QUESTIONS: What we need to know about our LOCAL PSTN telco (digital) lines to be shure in our hardware choise (voltage, current etc.)? We need 4 PSTN line system which can recive and make calls (on all four lines) from and to the local VoIP network, hardware sugestions PLEASE? Can four Digium Wildcard X100P make that happen and is that stabile in production?- we need to make working sistem and also cheap system? We are still learning Asterisk so sorry for this beginner mail. Thanks to all Darko ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell Hardware
We use SCSI on the Supermicro Dual Xeon and have no problems. I think I will see what happens when I use the IDE CDROM. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Friday, July 22, 2005 9:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell Hardware Lists wrote: > On Friday 22 July 2005 18:18, Andrea Cristofanini - Gedam Europe Srl > wrote: > >>We use Supermicro and we have NO problem at all :-) >> > > > Ditto, SuperMicro is solid. One of the best price/performance/quality > servers > you can buy. I'm still in therapy after my first (and last) experience with a SuperMicro Xeon board. HDLC aborts in a TE110P anytime there was IDE activity. Tried everything. setpci latency, hdparm for IDE settings, no graphics, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Business Edition
> On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: > > Adam Goryachev [EMAIL PROTECTED] wrote: > > > On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: > > > > For this reason, I believe that if a fork were > > > > ever necessary, it would struggle to beat a distinct path away from > > > > the Asterisk Binary Edition > > > > > > > Correct, until the point where there is MORE features being added to > > > the forked version of asterisk than the digium version of asterisk. > > > > > That can't happen, because the ABE could, and probably would, absorb > > all of the advances in the fork, while forging ahead with the > > original. > > > Since the fork would be GPL only, if ABE 'absorbed' the new features, > then it would 'become' GPL, and therefore would need to be released as > GPL, and hence would no longer by ABE :) So, that can't happen. Any other > ideas? > You're forgetting about the "disclaimer" documents. Anyone who signed the perpetual agreement and made changes and/or enhancements to the Asterisk code (a fork would still be using Asterisk code) would firstly be obliged to inform "the owner", and would secondly have a prior agreement with "the owner" to allow them to use and close the code. That would neatly bypass the GPL and allow the new code to be folded into the Asterisk Binary Edition. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
On Sun, Jul 24, 2005 at 12:43:06AM +1000, Adam Goryachev wrote: > On Sat, 2005-07-23 at 12:00 +0300, Tzafrir Cohen wrote: > > Disclaimers aside, who has the copyrights in those cases? > > Do you actually read the emails on this list? or just like to jump right > in and help the brawl continue? The disclaimers don't affect copyright, > the author of the work/patch/source code retains copyright That question was indeed mis-worded. But read the answer below it to see I refer to the ability to relicense, rather than to the copyrights themselves. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk users from Turkey?
Hi Ceyhan, We are specialized on Asterisk. Please check http://kulustur.com Please also see http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants for other consultants in Turkey, especially EMEA and Europe sections. I can try to answer your questions via email. But, people in this maillist have always been quite helpful. Soner - Original Message - From: "Ceyhun KIRMIZITAS" <[EMAIL PROTECTED]> To: Sent: Saturday, July 23, 2005 12:46 PM Subject: [Asterisk-Users] Asterisk users from Turkey? Is there any1 who has some experience with Asterisk in Turkey? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spa-2100 3.2.1 firmware
sorry to be a little off list, but I am provisioning sipura spa-2100 with firmware 3.2.1 Finally they got it working with fullduplex ethernet wan. The wan ethernet mode can be configured only by provisioning but I didn't find the provisioning field's name and the possible values. Is there some one that has just done it? thanks Rosario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Business Edition
Tzafrir Cohen [EMAIL PROTECTED] wrote: > Disclaimers aside, who has the copyrights in those cases? > > Digium currently holds copyrights and/or is allowed to relicense the > full asterisk codebase as is currently distributed in the asterisk > tarballs on ftp.asterisk.org and also all the code in the asterisk CVS > on cvs.digium.org (any better definition?) . > Just to be clear, the perpetual agreement doesn't force a transfer of copyright; The author gets to keep the copyright, and can do whatever he likes with the code. The shorter disclaimer puts the copyright into the public domain. The perpetual agreement gives "the owner" two main rights. Firstly paragraph 1 allows the code, and all future Asterisk-related code, written by that contributor to be closed by "the owner". Secondly, paragraphs 2 and 5(a) force the contributor to report all future changes and/or enhancements to save "the owner" the hassle of having to scour future forks looking for code that they might be interested in folding into their proprietary release. The second disclaimer (the short one) simply dumps all of your changes and enhancements into the public domain for anyone to use in a proprietary product. Of course, the only people who would know about this would be the signer and the company to which the document was sent. The short disclaimer is sufficiently woolly to allow for all future changes to a fork to be folded back into the binary release, although it doesn't include an obligation to report all such changes. Once signed, neither agreement has an exit clause or time limit, so neither of them can be cancelled. Maybe that is legal in Alabama (or Delaware), but I wouldn't really want to have to travel there to find out. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC gives me only the time, but no cost
The included seconds field is not taken into account when billing the
connect charge. IMHO this is a bug but I've not gotten enough feedback
to put the patch through. Therefore the patch has been closed. :-)
Darren Wiebe
[EMAIL PROTECTED]
Ronald Wiplinger wrote:
Darren Wiebe wrote:
The problem is in the line beginning with eval. It should read as
follows.
my $adjtime = eval { $adjtime = int((($answeredtime -
$numdata->{includedseconds}) + $increment - 1) / $increment) *
$increment;
return $adjtime };
Thanks, I solved it by removing "$adjtime = eval"
Are there other errors in the cvs verion?
bye
Ronald Wiplinger
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Re: [Asterisk-Users] Soft Phone
You could give SJphone a shot too.. http://www.sjlabs.com/sjp.html http://www.sjlabs.com/doc/SJphoneReadme.rtf <-- the README for SJphone On Fri, 2005-07-22 at 19:22, Time Bandit wrote: > > Can anyone recommend a good soft phone that's easy to configure under > > Asterisk and works well on a typical Windows XP system? > Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php > > If you want a SIP phone, try X-Lite : > http://www.xten.com/index.php?menu=products&smenu=xlite > > hth > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Mac OS X proves that it's easier to make UNIX pretty than it is to make Windows secure signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Business Edition
On Sat, 2005-07-23 at 12:00 +0300, Tzafrir Cohen wrote: > Disclaimers aside, who has the copyrights in those cases? Do you actually read the emails on this list? or just like to jump right in and help the brawl continue? The disclaimers don't affect copyright, the author of the work/patch/source code retains copyright Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Business Edition
On Fri, 2005-07-22 at 18:18 +0100, Kevin Walsh wrote: > Adam Goryachev [EMAIL PROTECTED] wrote: > > On Fri, 2005-07-22 at 04:15 +0100, Kevin Walsh wrote: > > > For this reason, I believe that if a fork were > > > ever necessary, it would struggle to beat a distinct path away from > > > the Asterisk Binary Edition > > Correct, until the point where there is MORE features being added to the > > forked version of asterisk than the digium version of asterisk. > That can't happen, because the ABE could, and probably would, absorb > all of the advances in the fork, while forging ahead with the > original. Since the fork would be GPL only, if ABE 'absorbed' the new features, then it would 'become' GPL, and therefore would need to be released as GPL, and hence would no longer by ABE :) So, that can't happen. Any other ideas? > > The *average* feeling of the community is that they are happy with > > the status quo. > The status quo has been disrupted with the unveiling of the Asterisk > Binary Edition. ABE was released ages ago (feels that way) with a small minority group who kicked up a stink and then promptly went back to their caves. Just as you are now kicking up a new stink, this will probably happen every few months, and hopefully I won't feel inclined to reply next time (nor will anybody else) as we all learn to just ignore the people who simply want to re-hash old non-arguments... You have had your say, you have heard all the people who agree with you (was there any?) and now we can all carry on as before. When you start your fork, feel free to announce it, and we shall see where it goes. (I suspect the same place as the MSN chatroom or forum or whatever it was for asterisk that was recently announced :) Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC gives me only the time, but no cost
Darren Wiebe wrote:
The problem is in the line beginning with eval. It should read as
follows.
my $adjtime = eval { $adjtime = int((($answeredtime -
$numdata->{includedseconds}) + $increment - 1) / $increment) *
$increment;
return $adjtime };
Thanks, I solved it by removing "$adjtime = eval"
Are there other errors in the cvs verion?
bye
Ronald Wiplinger
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[Asterisk-Users] astcc timestamps
The time stamps in ASTCC are useless as they are now: Fri Jul 22 15:06:25 2005 Wouldn't it be better to use something like: 2005-07-22 15:06:24 Fri I want to sort the records by date, but with the format now it is impossible... or do I miss something? bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and timeouts
On Sat, 2005-07-23 at 06:35 -0400, Joseph wrote: > >> exten => _6XXX,2,Busy > >> exten => _6XXX,3,Hangup > > But the whole point is that I don't want the caller to hear a busy > > signal or get hung up, I want the Queue to try the next available agent. > > Which it does at the moment, just with the errors mentioned in the error > > log file. > > Julian > This busy means, tell the queue app that the agent is busy. The queue > app willl go try someone else. The caller will keep hearing music. :) Julian, and others, If someone offers you a suggestion towards solving your problem, you might at least try it before discarding the advice. I (or the person replying to you) went to some effort to learn this stuff at some point, and to type the reply to you. [ok, so I deleted the rest of my rant...] Please, let's just get along, and try things out, even if we thing it won't work... PS, ok, if I suggest that rm -rf / will solve all your problems, then maybe you shouldn't try it :) Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC gives me only the time, but no cost
The problem is in the line beginning with eval. It should read as follows.
my $adjtime = eval { $adjtime = int((($answeredtime -
$numdata->{includedseconds}) + $increment - 1) / $increment) * $increment;
return $adjtime };
Darren Wiebe
[EMAIL PROTECTED]
Ronald Wiplinger wrote:
I try to track down an error that causes that Astcc just reports the
time, but not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment - 1) /
$increment) * $increment };
my $cost;
print STDERR "Adjusted time is $adjtime, cost is $adjcost with
$adjconn fee\n";
eval { $cost = int($adjcost * $adjtime / 60) };
$cost += $adjconn;
print STDERR "Total cost is $cost\n";
return $cost;
I have added changes some lines
} elsif ($res eq "ANSWER") {
-->my $increment = $carddata->{inc};
--xmy $cost = &calccost($adjconn, $adjcost,
$answeredtime, $increment);
-->print STDERR " adjconn is $adjconn\n adjcost is
$adjcost\n answeredtime is $answeredtime\n \
inc is $carddata->{inc}\n increment is $increment";
$carddata->{used} += $cost;
print STDERR "Total used is now $carddata->{used}\n";
&savecdr($cardno, $callerid, $phoneno, $trunk,
$res, $answeredtime, $cost,$callstart);
&savedata($carddata);
The result is:
[Jul 23 12:20:29] == Spawn extension (Line-optimize,
901185228376786, 512) exited non-zero on
'Local/[EMAIL PROTECTED],2'
res is ANSWER, answered time is 26
Adjusted time is , cost is 603 with 0 fee
Total cost is 0
adjconn is 0
adjcost is 603
answeredtime is 26
inc is 6
Total used is now 70
Why? What do I need to correct?
bye
Ronald Wiplinger
[Jul 23 12:20:29] -- AGI Script astcc.agi completed, returning 0
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[Asterisk-Users] Asterisk 1.2 is getting closer - please help
Dear Asterisk Community, Asterisk 1.0 was released at Astricon 2004, in September last year. It's been almost a year and we haven't been able to go ahead and release a new version. Now is the time to try to move forward again. As we've outlined before, the process is this: * Code freeze: At this point, we'll stop accepting new additions (new functions) to the source code. Bug fixes are more than welcome, but additions will be postponed until after release and added to the 1.3dev source code base (the new HEAD). * Release candidate: A release candidate will be produced as a tar.gz file on the FTP site. * Release of 1.2: The new release version of Asterisk, that replaces Asterisk 1.0 * Release of 1.2.1: The working version :-) of the new version of Asterisk So why 1.2.1? Well, the common feeling among developers is that "No one really tests anything until we release, so we will receive bug reports from the hour we release 1.2.0". Let's try to prove that they are wrong! What can you do to help this process? - * Set up a test system, and test CVS head in something that resembles your production environment. Scripts, phone, dialplan - make sure you use as many of the features as you can and use in production to make sure they work as expected in version 1.2 * Go wild and test at least two of the new features in 1.2 just for fun and make sure they work as documented. Or document how they work if it's not documented. Test the new realtime architecture, voicemail ODBC storage, AEL - the new scripting language, the new dialplan templates and constructs, the #exec config directive, attended transfers, native music on hold... The list is long. * If you have reported bugs or filed patches in the bugtracker (bugs.digium.com), make sure you reply quickly when a bug marshal or developer ask you questions or require more information. At this point, we're working very hard to clear out outstanding bugs and stabilize the additions that is waiting for inclusion in the CVS. We will close reports that we can't move forward if we do not get any responses. We can re-open later, but need to move forward. If we have a report of a proven bug that needs fixing, those will not be closed. Only unclear reports with no responses will be closed. * Visit the bug tracker at bugs.digium.com and help us test patches. Postitive and negative reports are both equally needed. There's no way a small team of core developers and bug marshals can test everything in there now. We need to decide which patches that are ready for inclusion, that are tested and documented. * If you find that we're missing documentation, please add to the readme files, write new ones. The Asterisk documentation team is ready to help you if you need assistance in this effort. * Disappoint the developers by making sure that the CVS head gets a thorough testing phase now, before release! * Update the Wiki on the 1.2 version. Make sure that you make it very clear that new features only work in 1.2 and releases after that so you won't confuse readers that use older versions. * Test Asterisk CVS head on other platforms than Linux: FreeBSD, OpenBSD, MacOS/X, Commodore VIC 20 - will it work? When is 1.2 scheduled to be released? - At usual with Open Source, we release when the software is ready for release. We do not release when it suits the marketing department, when we need a positive stock report or when customers require it. That said, we now are trying to focus on getting a release out of the door around September 1st. No promises, it all depends on your help and assistance to move forward. Please ask your boss for some time and resources to help the project with testing or dedicate resources within your company to help us. It's Open Source, meaning that everyone works together to make sure we get the software that works for our home, our company or our organization. Finding information --- If you have questions about the developer version, the base for the 1.2 release, use the #asterisk-dev channel on the freenode.net IRC. If you have questions about bug reports and patches, find a bug marshal in the #asterisk-bugs channel. To find out how to download or connect to the IRC channel, please visit http://www.asterisk.org Thank you for your assistance! /Olle - Astricon 2005 - With the Asterisk Solutions Showcase! * Conference, Exhibition, Developer's meeting, Tutorials * Anaheim California Oct 12-14 - http://www.astricon.net/2005/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/
RE: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware
Title: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware It's Digium, not Dell. I have two identical Dell 1850s, each with the allegedly offensive built-in E100 Ethernet ports. I placed a TE410P card in each. One worked great, the other would not modprobe. Upon examination, we discovered that the two TE410P cards had different firmware revisions. Turns out the one with the older version was the one that worked. We sent the second card back and they re-flashed it to the older version. It now works just fine. I had similar problems in a Dell 1750 with TDM400P. The Rev H card worked, but the Rev. I would not. From: [EMAIL PROTECTED] on behalf of M OSent: Sat 7/23/2005 2:34 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware Hello,Just chiming in here:>From: [EMAIL PROTECTED]>[mailto:[EMAIL PROTECTED]] On>Behalf Of>[EMAIL PROTECTED]>Sent: Viernes, 22 de Julio de 2005 01:24 p.m.>To: Asterisk Users Mailing List - Non-Commercial>Discussion>Subject: RE: [Asterisk-Users] Dell Hardware>> Mmhh nice !! So, why did Digium forbid it :)?>If Dell is so bad... why is a Dell 2850 server one of>the two listed on the compatibility list for ABE?>http://www.digium.com/index.php?menu=product_detail&category=software&produc>t=ABE&tab=compatibilityI am running Asterisk on a 100Mbps Pipe on thefollowing:Hardware InformationProcessors 4 <-(should be 2)Model Intel(R) Xeon(TM) CPU 3.00GHzChip MHz 2992.81 MHzCache Size 1024 KBSystem Bogomips 23907.52PCI Devices 00:1f.1 IDE interface: Intel Corp.82801EB/ER02:0e.0 RAID bus controller: Dell PowerEdge ExpandableRAID controller 406:07.0 Ethernet controller: Intel Corp. 82541GI/PIGigabit Ethernet Controller07:08.0 Ethernet controller: Intel Corp. 82541GI/PIGigabit Ethernet Controller0b:0d.0 VGA compatible controller: ATI TechnologiesInc Radeon RV100 QY [Radeon 7000/VE]IDE Devices hda: TEAC CD-ROM CD-224ESCSI Devices MegaRAID LD 0 RAID0 69G (Direct-Access)PE/PV 1x6 SCSI BP (Processor)USB Devices Linux 2.4.21-27.0.1.ELsmp ehci-hcd IntelCorp. 82801EB USB2 00:1d.7USB UHCI Root Hub bca0USB UHCI Root Hub bcc0Sincerely,Martin__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail passwords located in #include file
On Fri, Jul 22, 2005 at 04:35:23PM -0700, Darren Parko wrote: > Hello, > > I have an #include file containing user voicemail configurations. > This works fine for the most part, but when a user changes their > password via the phone the #include file is not updated. > > Is there a way to do this? I use the option to change the passwords through an external script (externpass) However by default when externpass is used, the internall passowrds are not saved. See http://bugs.debian.org/312770 . A patch is attached to the last message there. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk users from Turkey?
Is there any1 who has some experience with Asterisk in Turkey? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues and timeouts
Asterisk wrote:
I've got several agents on a queue. However, they often forget to go
"not ready" or log off when they can't answer the phone.
I would like a person calling my queue to be on the queue for a max
of 2 minutes, and I'm using the rrmemory strategy.
I put a timeout of 12 on the call to my agent in the [AgentQ] context
(they log on using Agentlogincallback). It all seems to work ok,
except that I get a load of "pbx.c: Timeout, but no rule 't' in
context 'AgentQ'" in the error log. What would I use in the 't' rule
to stop this error from ocurring ?
and that is ? (I'm very slow on the uptake ...)
/* extensions.conf portion for calling agent */
...
[AgentQ]
exten => _6XXX,1,Dial(SIP/${EXTEN},12)
...
You have ... so I don't know what that means, but perhaps you could add:
the ... means more lines of the extensions.conf before the lines in
question
exten => _6XXX,2,Busy
exten => _6XXX,3,Hangup
But the whole point is that I don't want the caller to hear a busy
signal or get hung up, I want the Queue to try the next available agent.
Which it does at the moment, just with the errors mentioned in the error
log file.
Julian
This busy means, tell the queue app that the agent is busy. The queue
app willl go try someone else. The caller will keep hearing music. :)
--
respectfully, Joseph
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Re: [Asterisk-Users] (cause 66 - Channel not implemented) -- IAX?
[EMAIL PROTECTED] wrote:
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Goto("Zap/1-1", "MainMenu|s|1") in new stack
-- Goto (MainMenu,s,1)
-- Executing BackGround("Zap/1-1", "Welcome") in new stack
-- Playing 'Welcome' (language 'en')
-- Executing BackGround("Zap/1-1", "En-MainMenu") in new stack
-- Playing 'En-MainMenu' (language 'en')
== CDR updated on Zap/1-1
-- Executing Queue("Zap/1-1", "callcenter") in new stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1'
-- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX/1001") in new stack
Jul 23 14:11:25 WARNING[5713]: channel.c:2180 ast_request: No channel
type registered for 'IAX'
Jul 23 14:11:25 NOTICE[5713]: app_dial.c:1092 dial_exec_full: Unable to
create channel of type 'IAX' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Called Agent/1001
-- Stopped music on hold on Zap/1-1
Does the error mean IAX channel is not implemented? When * starts up
with -vvvc I get this message when iax2 loads.
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Registered custom function IAXPEER
== Registered application 'IAX2Provision'
== Manager registered action IAXpeers
== Manager registered action IAXnetstats
== Parsing '/etc/asterisk/iax.conf': Found
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 16
== Binding IAX2 to default address 0.0.0.0:4569
== IAX Ready and Listening
== Loaded firmware 'iaxy.bin'
== Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
My configurations are at http://pastebin.ca/18329
I am using firefly as my iax client, and it does not seems to work when
I use 1001,1,Dial(IAX2/1001) instead of 1001,1,Dial(IAX/1001)
Any help in resolving this will be much appreciated.
Change the lines below from IAX to IAX2
Now you have:
46. exten => 1001,1,Dial(IAX/1001)
47. exten => 1002,1,Dial(IAX/1002)
48. exten => 1003,1,Dial(IAX/1003)
49. exten => 1004,1,Dial(IAX/1004)
But you need:
46. exten => 1001,1,Dial(IAX2/1001)
47. exten => 1002,1,Dial(IAX2/1002)
48. exten => 1003,1,Dial(IAX2/1003)
49. exten => 1004,1,Dial(IAX2/1004)
--
respectfully, Joseph
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[Asterisk-Users] ASTCC gives me only the time, but no cost
I try to track down an error that causes that Astcc just reports the time, but
not the costs.
I could narrow the problem down into this sub routine:
sub calccost() {
my ($adjconn, $adjcost, $answeredtime, $increment) = @_;
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment)
* $increment };
my $cost;
print STDERR "Adjusted time is $adjtime, cost is $adjcost with $adjconn
fee\n";
eval { $cost = int($adjcost * $adjtime / 60) };
$cost += $adjconn;
print STDERR "Total cost is $cost\n";
return $cost;
I have added changes some lines
} elsif ($res eq "ANSWER") {
-->my $increment = $carddata->{inc};
--xmy $cost = &calccost($adjconn, $adjcost, $answeredtime,
$increment);
-->print STDERR " adjconn is $adjconn\n adjcost is
$adjcost\n answeredtime is $answeredtime\n \
inc is $carddata->{inc}\n increment is $increment";
$carddata->{used} += $cost;
print STDERR "Total used is now $carddata->{used}\n";
&savecdr($cardno, $callerid, $phoneno, $trunk, $res,
$answeredtime, $cost,$callstart);
&savedata($carddata);
The result is:
[Jul 23 12:20:29] == Spawn extension (Line-optimize, 901185228376786, 512)
exited non-zero on 'Local/[EMAIL PROTECTED],2'
res is ANSWER, answered time is 26
Adjusted time is , cost is 603 with 0 fee
Total cost is 0
adjconn is 0
adjcost is 603
answeredtime is 26
inc is 6
Total used is now 70
Why? What do I need to correct?
bye
Ronald Wiplinger
[Jul 23 12:20:29] -- AGI Script astcc.agi completed, returning 0
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Re: [Asterisk-Users] X100P not answering
On Fri, Jul 22, 2005 at 05:39:43PM -0500, Carlos Chavez wrote:
> I have an Asterisk server running todays CVS (updated it just in case
> that was the problem). It has 3 X100P cards. The first two lines I use
> as my normal work lines and the third is my fax line which I use with
> SpanDSP. I run Fedora Core 4.
Interested to know how this works. There seem to be quite a few problems
for spandsp through a TDM card, right?
>
> I have a problem that the third X100P does not answer the call. From
> the console I can see that there is an incoming call with the following:
>
> -- Starting simple switch on 'Zap/3-1'
> Jul 22 17:34:21 ERROR[30490]: callerid.c:266 callerid_feed: fsk_serie
> made mylen < 0 (-16)
> Jul 22 17:34:21 WARNING[30490]: chan_zap.c:5827 ss_thread: CallerID feed
> failed: Success
> Jul 22 17:34:21 WARNING[30490]: chan_zap.c:5871 ss_thread: CallerID
> returned with error on channel 'Zap/3-1'
Any chance that this is the same issue as the UK caller ID issue? I seem
to recall that the same problem applies to Mexican phone lines as well.
> -- Executing Wait("Zap/3-1", "1") in new stack
> -- Executing Answer("Zap/3-1", "") in new stack
> -- Executing ResponseTimeout("Zap/3-1", "5") in new stack
> -- Set Response Timeout to 5
> -- Executing SetLanguage("Zap/3-1", "es") in new stack
> -- Executing AbsoluteTimeout("Zap/3-1", "600") in new stack
> -- Set Absolute Timeout to 600
> -- Executing BackGround("Zap/3-1", "bien-fax") in new stack
> -- Playing 'bien-fax' (language 'es')
> Jul 22 17:34:25 WARNING[30490]: chan_zap.c:3696 zt_handle_event:
> Ring/Off-hook in strange state 6 on channel 3
> == Spawn extension (fax, s, 6) exited non-zero on 'Zap/3-1'
> -- Hungup 'Zap/3-1'
>
> The calling phone just keeps ringing all the time and the messages
> above are repeated until I hangup the call. My other two lines work
> perfectly. Any ideas?
Need an Answer in the dialplan?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] RE: Business Edition
On Fri, Jul 22, 2005 at 10:56:42AM -0500, Kevin P. Fleming wrote: > Kevin Walsh wrote: > > >The perpetual agreement grants "the owner" a "non-cancellable right > >to use changes and/or enhancements" made to the Asterisk codebase "as > >[the] owner sees fit." As any Asterisk fork would, of course, be based > >upon existing Asterisk code, "the owner" would have the automatic right > >to take any code they wanted and backport it into the Asterisk Binary > >Edition - as long as the contributor to the fork had previously signed > >a perpetual "disclaimer" at some point in the past. > > Nice work clipping out only the words you wanted to use there! Let's try > this again, with the actual text from the disclaimer: > > (b) The rights made in Para. 1(a) of this Agreement applies to all past > and future contributions of Contributer that constitute changes and > enhancements to the Program. > > 2. Contributer shall report to Owner all changes and/or enhancements to > the Program which are covered by this Agreement, and (to the extent known > to Contributer) any outstanding rights, or claims of rights, of any > person, that might be adverse to the rights of Contributer or Owner. > > In other words, the _only_ code that the disclaimer covers is that which > the Contributer directly identifies to Digium to be covered by the > disclaimer. In absolutely no way does this disclaimer give Digium the > right to appropriate other changes the Contributer makes to the covered > programs without their knowledge and permission. > > In addition, even the most liberal interpretation of these clauses still > includes the words "Contributer" and "contribution", which clearly means > that the entity signing the disclaimer has sole discretion which of > their changes are covered and which are not. "Contribution" there does not mention the main Asterisk source tree. Paragraph 1(a) defines "Program" as: the programs Asterisk, Gnophone, Phonecore, libiax No reference is made regarding a specific source tree. But why is such an over-broad license needed in the first place? Suppose a certain Kevin wrote a a patch to Asterisk that implements chan_telapathy (a feature that was requested by his technical support center). Kevin wants that code to be distributed with the main Asterisk codebase so to reduce maintinance costs for that channel. It is reasonable to assume that during that maitinance the code of that channel will be changed. Maybe some of its code will be used in other parts of Asterisk. Kevin also sometimes finds a bug in parts of the Asterisk code that he did not write. In that case he submits a patch. Disclaimers aside, who has the copyrights in those cases? Digium currently holds copyrights and/or is allowed to relicense the full asterisk codebase as is currently distributed in the asterisk tarballs on ftp.asterisk.org and also all the code in the asterisk CVS on cvs.digium.org (any better definition?) . Assuming digium wishes to relicense that code (and that it is OK for Kevin), Kevin can permit Digium to relicense those two specific contributions. So Kevin should have permitted Digium to relicense the current versions he submitted. If Kevin ever writes a new version of chan_telepathy and still wants it included in Asterisk, he should simply permit Digium to relicense the new code. So: a. There should be no reason for Digium to require anything about future contributions. b. The program should be well defined. A public CVS is a relatively good definition, as it is easy for others to get and save snapshots of it (not to mention that -cvs mailing list), and thus it can't be easily changed in the future. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (cause 66 - Channel not implemented) -- IAX?
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Goto("Zap/1-1", "MainMenu|s|1") in new stack
-- Goto (MainMenu,s,1)
-- Executing BackGround("Zap/1-1", "Welcome") in new stack
-- Playing 'Welcome' (language 'en')
-- Executing BackGround("Zap/1-1", "En-MainMenu") in new stack
-- Playing 'En-MainMenu' (language 'en')
== CDR updated on Zap/1-1
-- Executing Queue("Zap/1-1", "callcenter") in new stack
-- Started music on hold, class 'default', on Zap/1-1
-- outgoing agentcall, to agent '1001', on 'Local/[EMAIL PROTECTED],1'
-- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX/1001") in new stack
Jul 23 14:11:25 WARNING[5713]: channel.c:2180 ast_request: No channel
type registered for 'IAX'
Jul 23 14:11:25 NOTICE[5713]: app_dial.c:1092 dial_exec_full: Unable to
create channel of type 'IAX' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Called Agent/1001
-- Stopped music on hold on Zap/1-1
Does the error mean IAX channel is not implemented? When * starts up
with -vvvc I get this message when iax2 loads.
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Registered custom function IAXPEER
== Registered application 'IAX2Provision'
== Manager registered action IAXpeers
== Manager registered action IAXnetstats
== Parsing '/etc/asterisk/iax.conf': Found
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 16
== Binding IAX2 to default address 0.0.0.0:4569
== IAX Ready and Listening
== Loaded firmware 'iaxy.bin'
== Parsing '/etc/asterisk/iaxprov.conf': Found
-- Loaded provisioning template 'default'
My configurations are at http://pastebin.ca/18329
I am using firefly as my iax client, and it does not seems to work when
I use 1001,1,Dial(IAX2/1001) instead of 1001,1,Dial(IAX/1001)
Any help in resolving this will be much appreciated.
regards,
raj
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Re: [Asterisk-Users] Soft Phone
U can try the firefly. this softphone can be used w/ more then 1 line... https://www.virbiage.com/download.php To configurate this in asterisk is like a normal sip phone or iax phone... -SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) videosupport=yes ; Support Video disallow=all ; Disable all codecs allow=alaw allow=ulaw allow=g726 allow=g723.1 allow=g729 allow=gsm context = sip ; Send unknown SIP callers to this context ; - SIP ACOUNT - [INT_10] type=friend qualify=no ;500 host=dynamic dtmfmode=inband callgroup=2 pickupgroup=2 callerid="Gianni" <10> secret=segredo BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opteron Hardware with Asterisk
Hi, asterisk compiled fine and is running very stable on our dual opteron in 64 bit mode. When loading G.729 library we have to peload libz manually for any reason, but besides that minor issue, everthing is fine. We didn't yet test the limits of that machine. Roger. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
