Re: [asterisk-users] 's' extension Asterisk 1.2.18
Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel running? -- Original Message -- From: OCOSA ListAcct [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sat, 14 Jul 2007 14:56:33 -0500 how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk ncurses dependencies
Hello, I'm deploying asterisk on a comtrend mips adsl router, I'm aware of the dependence of libncurses, so I compiled ncurses 5.6 for that platform, As you must Know this devices are not resource wide and flash memory especially, after ncurses compilation I have a /usr/share/terminfo with 1,6 MB space and some other libraries, libmenu libform llibpanel etc, I would like to know if some developers or other person with asterisk knowledge point me to the exact resources needed fom ncurses in order to delete everything else given that asterisk is the only soft is going to use ncurses here... thanks -- Francisco J. Pérez Botella ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDI behind NAT?
Is there any way to set the targeting ip that is sent out in the
dundi answer (to my public ip or any other where i want to receive the
call)?
Change your mapping in dundi.conf to reflect your true public IP rather than
using ${IPADDRESS}.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
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Re: [asterisk-users] 's' extension Asterisk 1.2.18
Yeah thats what I thought I found everything running so I just upgraded and fixed the problem. Otis Anthony Francis wrote: Maybe a hardware problem? What does zttool and ztcfg -vvv say? Is Zaptel running? -- Original Message -- From: OCOSA ListAcct [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sat, 14 Jul 2007 14:56:33 -0500 how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] surge protector?
APC makes a two line unit. PTEL2. But it's two lines in one jack. Another - www.ablecom.com is a bit more Pro Just do a google and take your pick. joe a. On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote: I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] surge protector?
Joe acquisto wrote: APC makes a two line unit. PTEL2. But it's two lines in one jack. I have always been a fan of Triplite. They use old tech when appropriate. I am big on Line Conditioners and UPSs with line conditioning. Of course power in my house is really bad. Anything big kicks on and lights flicker (high risers off of sags are worst, and much more common that spikes above norm). Of course you need Telco protection, not AC protection (well you need both). The Triplite gear that I have looked at had very simple Thiristor (sp?) dumps to ground and fast clamping. Another - www.ablecom.com is a bit more Pro Just do a google and take your pick. joe a. On 7/14/2007 at 10:17 PM, Todd H [EMAIL PROTECTED] wrote: I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with iax2 over satellite
Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with iax2 over satellite
I have a client that is using SIP over satellite with G729, VAD and Jitter buffer. The calls are coming in great. - Original Message - From: Tom Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Sunday, July 15, 2007 5:50 PM Subject: [asterisk-users] Asterisk with iax2 over satellite Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with iax2 over satellite
Sip would probably work well in a single phone situation, but what I'm trying to do is use multiple phones over a single trunk connection. Using sip though do you have a few seconds at the beginning of each call where the audio is not clear? On our link when I tried a sip phone the connection was unstable for about 5 seconds before the two people could hear each other well enough to have a conversation. Also another things I noticed is that with either sip or Iax that when I call came in the latency shot through the roof up to 2.5 seconds or so for the first 30 seconds of the call before it settled back down to the usual 600 to 700. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, July 15, 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with iax2 over satellite I have a client that is using SIP over satellite with G729, VAD and Jitter buffer. The calls are coming in great. - Original Message - From: Tom Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Sunday, July 15, 2007 5:50 PM Subject: [asterisk-users] Asterisk with iax2 over satellite Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with iax2 over satellite
Tom, The 2.5 second latency is probably what is causing it. I do not know how long it takes for the call to stabilize but I am sure that if it took 2.5 seconds that I would of heard about it already. You have to look in to why in the initial 30 seconds the latency is at 2.5 seconds. I know that my client has 600-700 ms latency up and down from the satellite and then another 150 to us. Do you have jitter buffer enabled ? Also if you do steady pings what happens ? I would also to a tcpdump at the clients side and see what is going out. Hope this helped. Dovid - Original Message - From: Tom Moore [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Sunday, July 15, 2007 7:37 PM Subject: Re: [asterisk-users] Asterisk with iax2 over satellite Sip would probably work well in a single phone situation, but what I'm trying to do is use multiple phones over a single trunk connection. Using sip though do you have a few seconds at the beginning of each call where the audio is not clear? On our link when I tried a sip phone the connection was unstable for about 5 seconds before the two people could hear each other well enough to have a conversation. Also another things I noticed is that with either sip or Iax that when I call came in the latency shot through the roof up to 2.5 seconds or so for the first 30 seconds of the call before it settled back down to the usual 600 to 700. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, July 15, 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with iax2 over satellite I have a client that is using SIP over satellite with G729, VAD and Jitter buffer. The calls are coming in great. - Original Message - From: Tom Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Sunday, July 15, 2007 5:50 PM Subject: [asterisk-users] Asterisk with iax2 over satellite Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] choppy sound when transcoding (after os update)
after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ‘ast_context_add_ignorepat2’: ael_main.c:306: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type ael_main.c: In function ‘ast_context_add_switch2’: ael_main.c:328: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type ael_main.c: In function ‘ast_context_add_include2’: ael_main.c:317: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type [CC] ast_expr2f.c - ast_expr2f.o ast_expr2.fl: In function ‘ast_yyerror’: ast_expr2.fl:376: warning: passing argument 1 of ‘expr2_token_subst’ discards qualifiers from pointer target type chan_agent.c: In function ‘__agent_start_monitoring’: chan_agent.c:393: warning: the address of ‘savecallsin’ will always evaluate as ‘true’ chan_agent.c:396: warning: the address of ‘urlprefix’ will always evaluate as ‘true’ [LD] chan_agent.o - chan_agent.so [CC] chan_iax2.c - chan_iax2.o chan_iax2.c: In function ‘iax2_prune_realtime’: chan_iax2.c:2050: warning: passing argument 1 of ‘expire_registry’ discards qualifiers from pointer target type ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.7 chan_alsa : snd_pcm_open failed
asterisk-1.4.7, Fedora 7, intel emt64 - nocona: == Parsing '/etc/asterisk/alsa.conf': Found ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to open slave [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365 alsa_card_init: snd_pcm_open failed: No such file or directory [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:481 soundcard_init: Problem opening alsa I/O devices == No sound card detected -- console channel will be unavailable But: ls /dev/snd -l total 0 crw-rw-rw-+ 1 root root 116, 7 2007-07-14 14:22 controlC0 crw-rw-rw-+ 1 root root 116, 6 2007-07-14 14:22 pcmC0D0c crw-rw-rw-+ 1 root root 116, 5 2007-07-14 14:22 pcmC0D0p crw-rw-rw-+ 1 root root 116, 4 2007-07-14 14:22 pcmC0D1p crw-rw-rw-+ 1 root root 116, 3 2007-07-14 14:22 seq crw-rw-rw-+ 1 root root 116, 2 2007-07-14 14:22 timer and alsamixer works. Card: HDA Intel Chip: SigmaTel STAC9220D/9223D A2 I haven't changed alsa.conf, I haven't set up .asoundrc. Any help appreciated. sean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Tzafrir Cohen schrieb: On Sat, Jul 14, 2007 at 01:23:35PM +0200, Christophorus Laube wrote: Hi list, I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, I found an old feature-request bug in Zaptel which seems relevant: http://bugs.digium.com/3554 Not sure if this means that the feature is supported. Maybe ask Mathew Fredrikson or Digium support. by the way: Is this call deflection or ECT etc. only possible to be executed at ring time or can I redirect a yet running call? Thanks, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow list
Doug Lytle wrote: Looks like mail is getting held up between INXS.digium.internal and lists.digium.com Here's what I get: ---cut--- Received: from lanai.amooma.com ([127.0.0.1]) by localhost (lanai.amooma.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id DYhd6TXxmG6B; Sun, 15 Jul 2007 20:29:32 +0200 (CEST) Received: from lists.digium.com (lists.digium.com [216.207.245.17]) by lanai.amooma.com (Postfix) with ESMTP id 4283B1823E; Sun, 15 Jul 2007 20:29:32 +0200 (CEST) Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I6VKF-0001wt-4r; Thu, 05 Jul 2007 12:41:07 -0500 Received: from exprod8mx73.postini.com ([64.18.3.173] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I6VK5-0001wU-CO for asterisk-users@lists.digium.com; Thu, 05 Jul 2007 12:40:57 -0500 Received: from source ([66.231.39.211]) by exprod8mx73.postini.com ([64.18.7.10]) with SMTP; Thu, 05 Jul 2007 13:40:55 EDT Received: from jk1281.epiinc.com ([192.168.145.1]) by drdos.info with hMailServer ; Thu, 5 Jul 2007 13:39:37 -0400 Received: from jk1281.epiinc.com ([64.136.253.80] helo=jk1281.epiinc.com) by ASSP-nospam; 5 Jul 2007 13:40:55 -0301 Message-ID: [EMAIL PROTECTED] ---cut--- Looks like lists.digium.com is the culprit. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Hi! I found an old feature-request bug in Zaptel which seems relevant: http://bugs.digium.com/3554 Not sure if this means that the feature is supported. Maybe ask Mathew Fredrikson or Digium support. by the way: Is this call deflection or ECT etc. only possible to be executed at ring time or can I redirect a yet running call? ECT only works after having answered a call (and having put it on hold on the ISDN side, not in Asterisk), see readme of chan_capi-cm for more info. Deflection, on the other hand, must be arranged before a call is answered. I add to the wiki a tiny overview for the different methods and channels: http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer Of interest for you are probably: * chan_misdn has this dialplan app: misdn_facility(calldeflect|) * ZapCD for deflection as implemented by bristuff Cheers, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
On Wed, 11 Jul 2007 19:57:05 +1000, Bill Maidment wrote On Wed, 4 Jul 2007 17:37:29 +0200, Christian Victor wrote I have the same problem. My mail sent yesterday around 20:00h and it still not arrived at the list. Sent from germany by the way. Christian email delays here are about 8 days. I don't expect to see this until 19th July -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.vu ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Finally received it (a litle earlier than I'd anticipated). Please note I'm located in Sydney, Australia so the time here is UTC+10.00 Here are the headers From [EMAIL PROTECTED] Wed Jul 11 19:57:05 2007 Return-Path: [EMAIL PROTECTED] Received: from mail2.maidment.vu (mail2.maidment.vu [192.168.2.6]) by mail1.maidment.vu (8.14.1/8.14.1) with ESMTP id l6FIWYjN013962 for [EMAIL PROTECTED]; Mon, 16 Jul 2007 04:32:34 +1000 Received: from lists.digium.com (lists.digium.com [216.207.245.17]) by mail2.maidment.vu (8.14.1/8.14.1) with ESMTP id l6FIWMYl019169 for [EMAIL PROTECTED]; Mon, 16 Jul 2007 04:32:29 +1000 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I8Ywl-0006nM-E0; Wed, 11 Jul 2007 04:57:23 -0500 Received: from exprod8mx66.postini.com ([64.18.3.166] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I8Ywd-0006nG-8i for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 04:57:15 -0500 Received: from source ([150.101.116.251]) by exprod8mx66.postini.com ([64.18.7.10]) with SMTP; Wed, 11 Jul 2007 02:57:13 PDT Received: from mail1.maidment.vu (mail1.maidment.vu [192.168.2.8]) by mail2.maidment.vu (8.14.1/8.14.1) with ESMTP id l6B9v8M8007038 for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 19:57:08 +1000 Received: from mail1.maidment.vu (localhost.localdomain [127.0.0.1]) by mail1.maidment.vu (8.14.1/8.14.1) with ESMTP id l6B9v5km020073 for asterisk-users@lists.digium.com; Wed, 11 Jul 2007 19:57:05 +1000 From: Bill Maidment [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 11 Jul 2007 19:57:05 +1000 Message-Id: [EMAIL PROTECTED] In-Reply-To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] X-Mailer: OpenWebMail 2.52 20060502 X-OriginatingIP: 192.168.2.119 (bill) MIME-Version: 1.0 X-Spam-Score: -2.6 () BAYES_00 from 192.168.2.8 X-Spam-Score: -2.6 () BAYES_00 from 192.168.2.6 X-Spam-Score: -4.4 () ALL_TRUSTED,BAYES_00 from 192.168.2.6 X-Spam-Score: -4.4 () ALL_TRUSTED,BAYES_00 from 192.168.2.8 Content-Disposition: inline X-Scanned-By: MIMEDefang 2.62 on 192.168.2.8 X-Scanned-By: MIMEDefang 2.62 on 150.101.116.251 X-Scanned-By: MIMEDefang 2.62 on 192.168.2.6 X-Scanned-By: MIMEDefang 2.62 on 192.168.2.8 X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S:66.28460/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] Subject: Re: [asterisk-users] List delays X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.9 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Status: R -- Bill Maidment Maidment Enterprises Pty Ltd www.maidment.vu ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking Valet
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/ Indicates support for Asterisk 1.4. The documentation listed suggests an install like so: cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c However astxs doesn't seem to be present in asterisk 1.4 Does anyone have this working with 1.4? and any suggestions on how to install? Thanks Kevin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trials with 1.4
Hi Ira - In the end my issue would seem to be I did something out of order, though I thought I did it right, and as much as I tried the software only gave meaningless to me messages. I didn't understand and I don't think I've seen it said before that I should run .\configure every time I do anything. I didn't understand what it did and had no idea it was how I'd get Zaptel support to build. You shouldn't need to run the configure script every time you do something. Here's a few provisos to help everything go smoothly: 1. The configure script for Zaptel is not the same as the configure script for Asterisk. You'll need to run both if you want to use both. 2. The accepted order to installing zaptel and asterisk: Zaptel first, then Asterisk 3. The asterisk configure script needs to figure out whether or not you want zaptel support (asterisk does not need zaptel), and it can only figure that out after you've run the zaptel configure script As far as ease of installation on 1.2 vs 1.4: Yes, if you use the make menuselect options there are a few more steps to install 1.4. These extra steps are optional and are there so you can avoid having to hand edit makefiles or manually enter command-line options. For many 1.2 installs, rather than do any manual editing, people just used the default files and ended up running too many zaptel drivers and asterisk modules. In 1.4, the menuselect options help you trim down your zaptel and asterisk installs to keep things lean. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
