Re: [asterisk-users] CallerID shows wrong values in manager interface
Thanks all :) Appreciate it. On Feb 1, 2008 12:04 PM, Ex Vito [EMAIL PROTECTED] wrote: I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- 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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ -- 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] call log notice messages
I am getting quite a lot of these notices: Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel 'mISDN/3-2' not posted Feb 1 10:22:01 NOTICE[2487] cdr.c: CDR on channel 'mISDN/3-2' lacks end Feb 1 10:22:06 NOTICE[2471] cdr.c: CDR on channel 'mISDN/3-1' not posted Feb 1 10:22:06 NOTICE[2471] cdr.c: CDR on channel 'mISDN/3-1' lacks end Feb 1 10:22:17 NOTICE[1591] cdr.c: CDR on channel 'mISDN/4-1' not posted Feb 1 10:22:17 NOTICE[1591] cdr.c: CDR on channel 'mISDN/4-1' lacks end Feb 1 10:23:40 NOTICE[2744] cdr.c: CDR on channel 'mISDN/3-1' not posted Feb 1 10:23:40 NOTICE[2744] cdr.c: CDR on channel 'mISDN/3-1' lacks end Feb 1 10:23:41 NOTICE[2157] cdr.c: CDR on channel 'mISDN/2-1' not posted Feb 1 10:23:41 NOTICE[2157] cdr.c: CDR on channel 'mISDN/2-1' lacks end Feb 1 10:23:51 NOTICE[2386] cdr.c: CDR on channel 'Zap/15-1' not posted Feb 1 10:23:51 NOTICE[2386] cdr.c: CDR on channel 'Zap/15-1' lacks end Feb 1 10:25:44 NOTICE[2516] cdr.c: CDR on channel 'mISDN/2-2' not posted Feb 1 10:25:44 NOTICE[2516] cdr.c: CDR on channel 'mISDN/2-2' lacks end Feb 1 10:26:23 NOTICE[3064] cdr.c: CDR on channel 'mISDN/4-1' not posted Feb 1 10:26:23 NOTICE[3064] cdr.c: CDR on channel 'mISDN/4-1' lacks end Feb 1 10:26:38 NOTICE[3086] cdr.c: CDR on channel 'SIP/4052-b519c460' not posted Feb 1 10:26:38 NOTICE[3086] cdr.c: CDR on channel 'SIP/4052-b519c460' lacks end Feb 1 10:26:50 NOTICE[3187] cdr.c: CDR on channel 'mISDN/4-2' not posted Feb 1 10:26:50 NOTICE[3187] cdr.c: CDR on channel 'mISDN/4-2' lacks end Feb 1 10:27:04 NOTICE[2857] cdr.c: CDR on channel 'mISDN/3-2' not posted Feb 1 10:27:04 NOTICE[2857] cdr.c: CDR on channel 'mISDN/3-2' lacks end What do they mean? Should I worry about call details not being correctly recorded? Vieri Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] meetme music on hold - when only conference member problem
Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) Kind regards tomasz ___ -- 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] h priority problem
You can use account code and userfield. You can set userfield to anything you want in the dialplan. On Feb 1, 2008 3:31 PM, Doug Lytle [EMAIL PROTECTED] wrote: Paul Hales wrote: Anyone have any ideas? What can I use to carry a variable over into 'h'?? Lets see what you have so far. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h priority problem
Paul Hales wrote: Anyone have any ideas? What can I use to carry a variable over into 'h'?? Lets see what you have so far. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme music on hold - when only conference member problem
On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) You probably don't have a timing source. If you don't have any telephony hardware installed you'll need the ztdummy module... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h priority problem
On 2/1/08, Paul Hales [EMAIL PROTECTED] wrote:
I need to carry a variable over into the 'h' priority - so I can go back
and clean up DB entries in a mysql database (time of call and so on)
I tried using UNIQUEID but it seems that 'h' generates a new one.
Anyone have any ideas? What can I use to carry a variable over into
'h'??
You have to understand that there's several channels for one call.
Variables are only inherited from parent channel to child, if you make
them inheritable by prepending underscore or two underscores. For
example - if you want to have one common unique identifier for call -
do this at beginning of each channel:
if (${call_id}=) {
Set(__call_id=${UNIQUEID});
}
Then you can use this to store something shared in asterisk internal DB()
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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[asterisk-users] Unicall
Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance ___ -- 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] play promt at the same time to calling and callee
Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten = s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten = s,2,Dial(SIP/trunk-out/37052390920|60|rL(10)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt asterisk plays hello world promt. -- SIP/trunk-out-08155880 Playing 'conf-enteringno' (language 'en') -- SIP/sip3.call.lt-08151550 Playing 'hello-world' (language 'en') So my question is , how to do this in the same time. Maybe somebody is using Dial G(context^exten^pri) for this purpose? Thanks ___ -- 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-Addons install success-Could not find ooh323.conf
[EMAIL PROTECTED] wrote: Am I doing something wrong? What I should do to get ooh323.conf cp asterisk-ooh323c/h323.conf.sample /etc/asterisk/ooh323.conf -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play promt at the same time to calling and callee
2008/2/1, Giedrius Augys [EMAIL PROTECTED]: Hello, I want that, when call is answered , callee and calling would hear different prompts and after promts the calls would be bridged. I've tried this situation: exten = s,1,Set(LIMIT_CONNECT_FILE=hello-world) exten = s,2,Dial(SIP/trunk-out/37052390920|60|rL(10)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt asterisk plays hello world promt. -- SIP/trunk-out-08155880 Playing 'conf-enteringno' (language 'en') -- SIP/sip3.call.lt-08151550 Playing 'hello-world' (language 'en') So my question is , how to do this in the same time. Maybe somebody is using Dial G(context^exten^pri) for this purpose? Thanks I have tried this : exten = s,1,Dial(SIP/trunk-out/37052390920|60|rG(music-testinis^s^1)) [music-testinis] exten = s,1,goto(1,1) exten = s,2,goto(2,1) exten = 1,1,Playback(lt/conf-enteringno) exten = 2,1,Playback(lt/conf-enteringno) but I get this: god*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/sip3.call.lt-08141e00, testuojame|s|1) in new stack -- Goto (testuojame,s,1) -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sip3.call.lt-08141e00, SIP/trunk-out/37052390920|60|rG(music-testinis^s^1)) in new stack -- Called trunk-out/37052390920 -- SIP/trunk-out-0818fb40 is ringing -- SIP/trunk-out-0818fb40 is making progress passing it to SIP/sip3.call.lt-08141e00 -- SIP/trunk-out-0818fb40 is making progress passing it to SIP/sip3.call.lt-08141e00 -- SIP/trunk-out-0818fb40 answered SIP/sip3.call.lt-08141e00 -- Executing [EMAIL PROTECTED]:1] Goto(SIP/sip3.call.lt-08141e00, 1|1) in new stack -- Goto (music-testinis,1,1) -- Executing [EMAIL PROTECTED]:1] Playback(SIP/sip3.call.lt-08141e00, lt/conf-enteringno) in new stack -- SIP/sip3.call.lt-08141e00 Playing 'lt/conf-enteringno' (language 'en') -- Executing [EMAIL PROTECTED]:2] Goto(SIP/trunk-out-0818fb40, 2|1) in new stack -- Goto (music-testinis,2,1) -- Executing [EMAIL PROTECTED]:1] Playback(SIP/trunk-out-0818fb40, lt/conf-enteringno) in new stack -- SIP/trunk-out-0818fb40 Playing 'lt/conf-enteringno' (language 'en') == Auto fallthrough, channel 'SIP/sip3.call.lt-08141e00' status is 'UNKNOWN' == Auto fallthrough, channel 'SIP/trunk-out-0818fb40' status is 'UNKNOWN' My question is , how to bridge these two calls. I'm using Asterisk 1.4.11, Thanks ___ -- 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] BRI card with PCI-E interface
On Fri, 2008-02-01 at 15:01 +0100, Alberto Pastore wrote: Olivier ha scritto: Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sangoma A500BRX is a 2 to 6 bri pci-x interface, although I've never tested it (A500BRECX comes with hw echo cancellation). We've been using the Sangoma A104DE in production for almost a year now and it works great. Bob ___ -- 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] realtime warning
yes. On Feb 1, 2008 12:07 PM, Russell Bryant [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Hi, The server log shows the following message. [Jan 29 04:59:02] WARNING[1896] config.c: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available Does it mean the server failed to file the mysql server? I have installed mysql and both asterisk and mysql are located in the same server. What do the message mean? It seems the message will cause the user failed to login. How can it be solved? Did you install res_config_mysql from asterisk-addons? -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Enterprise or Fedora?
i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/___ -- 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] BRI card with PCI-E interface
Olivier ha scritto: Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sangoma A500BRX is a 2 to 6 bri pci-x interface, although I've never tested it (A500BRECX comes with hw echo cancellation). Regards, Alberto. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- 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] BRI card with PCI-E interface
Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with DTMF dialing
On Thursday 31 January 2008 11:52:09 pm Ian wrote: Sorry for taking so long to reply, This email got lost in translation, again. Ian Ian said the following on 30-Jan-08 03:57 PM Thaks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 did you use zaptel-1.4.7 prior to this? did it work then? if so, it may be related to http://bugs.digium.com/view.php?id=11855 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year old. It is also considered a bleeding-edge distribution and is a testing ground for new applications and features. As such, it has frequent updates (including the kernel) that aren't guaranteed not to break your system. If stability and long-term support are your goals, I recommend taking a look at CentOS http://www.centos.org/. It is a binary-compatible clone of Red Hat Enterprise Linux that's free and has a very long support period. It's basically RHEL without the paid support contract. My migration to CentOS was painless, because the file system and configuration are practically identical to those of Fedora. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Matthew J. Roth a écrit : [...] I settled on using an empty TDM400P as a timing source, because it is a simple solution that just works. This may still be your best bet, but I'll defer judgment on that to the list because Asterisk has evolved quite a bit since I made that decision. This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme music on hold - when only conference member problem
I have, I have ztdummy module loaded in the kernel On Feb 1, 2008 11:59 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 1 Feb 2008, Tomasz Zieleniewski wrote: Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to invoke exten = s,n,MeetMe(|cdIMps) You probably don't have a timing source. If you don't have any telephony hardware installed you'll need the ztdummy module... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] AgentLogin by console
Thanks Tzafir, but this functionality needs sombody answer the call. I need to do this automatically. On Jan 22, 2008 4:10 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Jan 22, 2008 at 04:04:29PM -0200, equis software wrote: Hi! Is there any way to login an agent by console command? I want to login an agent doing this system call. asterisk -rx 'AgentCallbackLogin 304 [EMAIL PROTECTED]' Any ideas, thanks. Something of the sort of: originate SIP/peer-to-login application AgentCallbackLogin -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Enterprise or Fedora?
We are using Fedora because that is what the company we got our system from recommended. If I was doing the system myself I would throw in my vote for CentOS. I am using it for a database server and I have had no problems with it at all. It is about as stable and secure of Linux distro as I have ever used. If you do go with them I suggest kicking the CentOS team a few dollars. They do dang good work. On Feb 1, 2008 9:16 AM, love U. all [EMAIL PROTECTED] wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference Express yourself instantly with MSN Messenger! MSN Messenger ___ -- 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] SIP Softphones and Citrix ?
Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of cisco softphones with Citrix will come into play with SIP softphones as well. Seems that the two biggest issues revolve around wrapping the UDP stream up with the ICA protocol, and possibly issues with the various mics and speakers and having to interface with them I think. However, I am also a firm believer that anything is possible, practical well not usually, and it may just be the time has not come yet for this. There is a good article about this over at: http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server Any thoughts, comments or insight into this and your experiences around any of this is appreciated. ___ -- 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] Unicall
Mark, you are confusing terms here. You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that convention for its versions. What you have is astunicall-1.4.9-01. AstUnicall is just a package with patches and proper versions of spandsp, libsupertone, libunicall, libmfcr2, zaptel and Asterisk to run MFC/R2. Now, having said that, to answer your question, if you just recompile Asterisk w/o upgrading, then no, you don't need anything. If you want to upgrade Asterisk, then it depends. Small version upgrades (like 1.4.9 to 1.4.17) probably do not require to change anything, just copy channels/chan_unicall.c and the Makefile entries of channels/Makefile and you have good chances of being fine. Upgrading from 1.4 to 1.6 increases the chances of a broken compilation or runtime error. In general, if you have NO knowledge of C, then you will have to try and find yourself if it works or not :) Regards, Moisés Silva On Feb 1, 2008 7:24 AM, Mark Welch [EMAIL PROTECTED] wrote: Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] Remote Call Center Agents and Asterisk?
Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simply playing around. My background however is with Cisco CallManager, Cisco IPCCX for call centers as well as a mixed bag of other big name systems. I am simply researching and investigating different possibilites and solutions for a project at this point. Pursuing as many avenues as possible and trying to setup various test beds and labs if you will to accomplish the goal of one day rolling out home based remote call center agents. Look forward to hearing from others about this. Looking to hear of any success stories, as well not so successful stories. Trials and tribulations, good and bad experiences and where that left you. I know others have at the least done exactly what I am doing and have researched and entertained various ideas regarding this model of home based agents so hopefully this message can be a catalyst for further disscussion around this trend. ___ -- 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] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)
That sounds like a CANCEL message being improperly routed somewhere along the line. Nothing you can do with the config to fix that. N. Paul Madley wrote: Hi, Does anyone of you has a working configuration with SNOM phones that are able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our previous 1.4.13 installation. It also seems to manifest itself in ghost ringing as I've called it; place a call to a SIP extension, then put down the handset, yet the destination handset continues to ring out. Sorry I can't be of more use; if anybody else can suggest a workaround it'd be greatly appreciated! Thanks, P. ___ -- 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] Unicall
Thank you Moisés, we are indeed going from 1.4.9 to 1.4.17, we will backup channels/chan_unicall.c and the Makefile entries of channels/Makefile and do our upgrade to .17. You are indeed correct on the Unicall, we have astunicall-1.4.9-0.1, I thought it was the same thing. Now I know better :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Friday, February 01, 2008 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unicall Mark, you are confusing terms here. You DO NOT have Unicall 1.4.9-0.1, libunicall does not even use that convention for its versions. What you have is astunicall-1.4.9-01. AstUnicall is just a package with patches and proper versions of spandsp, libsupertone, libunicall, libmfcr2, zaptel and Asterisk to run MFC/R2. Now, having said that, to answer your question, if you just recompile Asterisk w/o upgrading, then no, you don't need anything. If you want to upgrade Asterisk, then it depends. Small version upgrades (like 1.4.9 to 1.4.17) probably do not require to change anything, just copy channels/chan_unicall.c and the Makefile entries of channels/Makefile and you have good chances of being fine. Upgrading from 1.4 to 1.6 increases the chances of a broken compilation or runtime error. In general, if you have NO knowledge of C, then you will have to try and find yourself if it works or not :) Regards, Moisés Silva On Feb 1, 2008 7:24 AM, Mark Welch [EMAIL PROTECTED] wrote: Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] G729 version to be downloaded for my machines
Download for Pentium4 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, January 30, 2008 10:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 version to be downloaded for my machines Hi List; The output of cat /proc/cpuinfo giving a [Intel (R) Pentium (R) D] so what is the g729 version I have to download to work with my machine? Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Meetme voice quality problems
Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Do you have any idea why? Can't the B410P be used as a timing source? What have you done to provide stable timing? I know that's a lot of questions, but I'm genuinely curious. It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)
Hi, Does anyone of you has a working configuration with SNOM phones that are able to pickup a call from a flasing LED? Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 release, and therefore I don't think any config changes will fix it. We've been told to roll back to our previous 1.4.13 installation. It also seems to manifest itself in ghost ringing as I've called it; place a call to a SIP extension, then put down the handset, yet the destination handset continues to ring out. Sorry I can't be of more use; if anybody else can suggest a workaround it'd be greatly appreciated! Thanks, P. ___ -- 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] Enterprise or Fedora?
Matthew J. Roth wrote: love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year I'd take it a step further... if you want asterisk on a stable platform, I'd suggest trixbox. It is CentOS with an additional asterisk repo. It also comes with several additional apps like SugarCRM and web based management. -- Milton Calnek BSc, A/Slt(Ret.) [EMAIL PROTECTED] 306-717-8737 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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's DANGEROUS Transfer CDR's
Hello, For such cases we usually suggest to put 2 boxes in your infrastructure: 1. Main billing gateway - where all PBX'es are connected (all client's remote PBX'es and your Local PBX) 2. Local PBX - where user's without PBX'es are connected Then user connects in following way: User - Local PBX - Main GTW - PSTN That way you will be save from transfer issue and all your clients will be able to transfer their calls on Local PBX. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: Wednesday, January 30, 2008 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's - Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 9:24:14 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's The provider can disable transfers (which is what we do), but why can a PBX not still allow it? Our PBX customers all can do transferring... but that's because billing isn't needed THERE. The billing, if any, is done on our end, or their providers end. This really seems like a very small and moot point that is being blown up. Depends how much it could cost you I guess :). If you're not supporting transfers it's a moot point if you are it's a bit more interesting. If the receptionist needs to transfer the call, then she should be able to do that within the confines of her PBX... the transfer of her call should NEVER go back out her PBX back to the supplier, for if it does, her PBX now loses control of that call. Our customer base is residential and small business. They don't want to either pay for or support another a PBX thats what they've come to us for in the first place a lot of the time. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] pulling my hair out over voicemail
Ok, I have made some progress debugging this. I dont believe it has anything to do with asterisk or my phone. Rather I think it is an issues with STUN and/or my Linksys router at home. The phones I am testing all sit behind a NAT'd firewall, your basic Linksys router for the Home DSL user. The phones all of STUN setup, and the STUN server IP is the IP of the asterisk server - which is purely public. I was able to duplicate the problem with not being able to hear the voicemail greeting by doing the following: Turn off all the phones, and power cycle my Linksys, then turn on 1 phone. That one phone will then work, and you can hear voicemail greeting. The I turn on the second phone. Then voicemail greeting breaks, and you cant hear it when you dial into voicemail. If I unplug the first phone, and power cycle the Linksys again, the second phone will begin to work. So the question is, does this behavior make sense? I assumed with an STUN server I could have multiple phones behind my Linksys firewall, now it appears I can only have one. Is it a Linksys bug, or a general known issue? Do I need to run multiple STUN servers? Thanks John On Jan 31, 2008, at 1:00 PM, Shane D wrote: Very odd. Could you try taking the mailbox line out of sip.conf and see what happens? On 1/31/08, John Von Essen [EMAIL PROTECTED] wrote: Here are my configs: sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 context=default extensions.conf: [default] exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain Calling from phone to phone is fine, and inbound and outbound calling is fine. But when I call voicemail, I dont hear anything. When I view console in CLI I see this when attempting to dial the voicemail extension: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081d65c8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081d65c8, 2) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081d65c8, [EMAIL PROTECTED]) in new stack -- SIP/6001-081d65c8 Playing 'vm-login' (language 'en') [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: Couldn't read username Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE So it plays the greetings, and is working, I just cant hear it. -john On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote: On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL PROTECTED] wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: Maybe the SIP config is wrong? Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Can you places other calls from that new phone? Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. What version? ___ -- 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 -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ -- 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] Codec preference selection, codec negotiation
Hi List; Is it possible to do configuration at the user context to let him use codec1 if destination support and if not, then use codec 2? For example, to let user1 use codec g729 if he needs to call user2 and user2 support g729, and if user2 does not support g729 then use g711 (alaw or ulaw), is it possible? I think this kind of settings required codec negotiation and I do not know if Asterisk has such capability. Any help? Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Enterprise or Fedora?
My suggestion - use the distro which you know best. We use Debian (200+ installations). It works stable for us because we know how to achieve it. Others use Fedora/Centos - because they are experts in these systems. Stability and performance of the system does not depend on the distro - only on person who built this system. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LWATCDR Sent: Friday, February 01, 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Enterprise or Fedora? We are using Fedora because that is what the company we got our system from recommended. If I was doing the system myself I would throw in my vote for CentOS. I am using it for a database server and I have had no problems with it at all. It is about as stable and secure of Linux distro as I have ever used. If you do go with them I suggest kicking the CentOS team a few dollars. They do dang good work. On Feb 1, 2008 9:16 AM, love U. all [EMAIL PROTECTED] wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference Express yourself instantly with MSN Messenger! MSN Messenger ___ -- 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
Re: [asterisk-users] Remote Call Center Agents and Asterisk?
On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote: Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simply playing around. My background however is with Cisco CallManager, Cisco IPCCX for call centers as well as a mixed bag of other big name systems. I am simply researching and investigating different possibilites and solutions for a project at this point. Pursuing as many avenues as possible and trying to setup various test beds and labs if you will to accomplish the goal of one day rolling out home based remote call center agents. Look forward to hearing from others about this. Looking to hear of any success stories, as well not so successful stories. Trials and tribulations, good and bad experiences and where that left you. I know others have at the least done exactly what I am doing and have researched and entertained various ideas regarding this model of home based agents so hopefully this message can be a catalyst for further disscussion around this trend. I have had several very successful implementations. Some small ~50-100 agents, and some larger, around 500 agents. The trick is keeping the agents honest since there is no supervisor standing behind them. You want to establish a minimum standard for the home agent. Recording of calls for sound quality and agent evaluation will be critical for QA as well as ChanSpy and whisper coaching which I believe is available in 1.4 (?) Use AJAX and Jabber ActiveX controls to control your CRM (web based of course). You could even ship them a Kit which contains a router running DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data center, an ATA, and a headset. There are several benefits to sending the router. Obviously, VPN. Then you also have something to SSH into and do testing like ping, traceroute, test throughput. You could even create some kind of app (if it doesn't already exist) to regularly run these diagnostics and upload them to you. That is just a few ideas but I think the main thing you will run into is agents not logging off or somehow trying to beat the system. That is what I see time and time again. Thanks, Steve Totaro ___ -- 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] Remote Call Center Agents and Asterisk?
On Feb 1, 2008 10:33 AM, d4rk f1br [EMAIL PROTECTED] wrote: Anyone using Asterisk in a Call Center environment? And more importantly is anyone supporting home based remote call center agents with an Asterisk backend? My experience with Asterisk is limited, however I have set it up and installed it previously and had it working for home usage and for simply playing around. My background however is with Cisco CallManager, Cisco IPCCX for call centers as well as a mixed bag of other big name systems. I am simply researching and investigating different possibilites and solutions for a project at this point. Pursuing as many avenues as possible and trying to setup various test beds and labs if you will to accomplish the goal of one day rolling out home based remote call center agents. Look forward to hearing from others about this. Looking to hear of any success stories, as well not so successful stories. Trials and tribulations, good and bad experiences and where that left you. I know others have at the least done exactly what I am doing and have researched and entertained various ideas regarding this model of home based agents so hopefully this message can be a catalyst for further disscussion around this trend. I have had several very successful implementations. Some small ~50-100 agents, and some larger, around 500 agents. The trick is keeping the agents honest since there is no supervisor standing behind them. You want to establish a minimum standard for the home agent. Recording of calls for sound quality and agent evaluation will be critical for QA as well as ChanSpy and whisper coaching which I believe is available in 1.4 (?) Use AJAX and Jabber ActiveX controls to control your CRM (web based of course). You could even ship them a Kit which contains a router running DD-WRT or OpenWRT setup with an OpenVPN tunnel back to your data center, an ATA, and a headset. There are several benefits to sending the router. Obviously, VPN. Then you also have something to SSH into and do testing like ping, traceroute, test throughput. You could even create some kind of app (if it doesn't already exist) to regularly run these diagnostics and upload them to you. That is just a few ideas but I think the main thing you will run into is agents not logging off or somehow trying to beat the system. That is what I see time and time again. Thanks, Steve Totaro ___ -- 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] Remote Call Center Agents and Asterisk?
Hello Steve, You are right on track and this is also what we have done with pretty good results. Of course now with Flex/Air there are a number of ways to enhance the service for the Customer/Agent Ed Mail: edpimentl[at]gmail.com Voip: edpimentl [SKype | GoogleTalk ] http://agileoss.com (Web2.0 and SOA Development ) http://mobiquity.ws (Private Label Social Network) http://youbiquity.ws (Power of One for all Social Networks) ___ -- 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] CallerID shows wrong values in manager interface
Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust kidding... -- but at http://www.astsee.com/ you can download the source code to my AstSee project -- it may provide some insight into what needs to be (or CAN be) gleaned from asterisk. I struggled with all this a year or more ago and manged to get it to work fairly as expected. A problem you might notice is that I believe I mistakenly update my internal arrays before asterisk's manager interface has sent a complete packet... argh But you can see me dealing with NewState, NewExten, NewChannel, etc etc and what I do with them :) Moj Devraj Mukherjee wrote: Thanks all :) Appreciate it. On Feb 1, 2008 12:04 PM, Ex Vito [EMAIL PROTECTED] wrote: I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- 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] Enterprise or Fedora?
On Fri, 1 Feb 2008, Milton Calnek wrote: Matthew J. Roth wrote: love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year I'd take it a step further... if you want asterisk on a stable platform, I'd suggest trixbox. It is CentOS with an additional asterisk repo. It also comes with several additional apps like SugarCRM and web based management. But all that cruft is not always appropriate -- parts left out don't get broke. They also don't need to be updated, suck up memory, cpu, disk or get in the way of learning how Asterisk really works. I like to start with a minimal CentOS install -- de-select everything and then yum install just the packages you actually need. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Astersik Transcoder support
Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- 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] Astersik Transcoder support
http://www.digium.com/en/products/voice/tc400b.php Simon Elliston Ball [EMAIL PROTECTED] On 1 Feb 2008, at 17:29, Charles Feng wrote: Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Never miss a thing. Make Yahoo your homepage. ___ -- 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] Enterprise or Fedora?
On Fri, Feb 01, 2008 at 09:58:28AM -0600, Milton Calnek wrote: Matthew J. Roth wrote: love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year I'd take it a step further... if you want asterisk on a stable platform, I'd suggest trixbox. It is CentOS with an additional asterisk repo. It also comes with several additional apps like SugarCRM and web based management. I'm sorry, but Trixbox (CE) and its like are modified versions of CentOS, and those modifications are not always so standard: 1. Many changes in /usr/local and such rather than in packages. 2. Kernel never gets updated. Even latest trixbox 2.2 versions are shipped with kernel from CentOS 4.3. CentOS has issued many updates since. Try to build your own kernel stuff on it. 3. It uses FreePBX, and this has its pros and cons. It is surely a comlpicated system to start with. If you happen not to like it, you're heading for much work in removing it. So it's not simply CentOS. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.17 and Teliax DTMF
I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended for DTMF and Teliax? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. -Original Message- From: Matthew J. Roth [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 7:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Enterprise or Fedora? love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year old. It is also considered a bleeding-edge distribution and is a testing ground for new applications and features. As such, it has frequent updates (including the kernel) that aren't guaranteed not to break your system. If stability and long-term support are your goals, I recommend taking a look at CentOS http://www.centos.org/. It is a binary-compatible clone of Red Hat Enterprise Linux that's free and has a very long support period. It's basically RHEL without the paid support contract. My migration to CentOS was painless, because the file system and configuration are practically identical to those of Fedora. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bypassing a Auth on Invite or Forbiden?
Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting each others IP 100%. But the PBX02 challenges PBX01's requests all the time even though insecure=invite is set. Here are the stanzas for each server: PBX01 [pbx01topbx02] type=friend context=incomingDefault host=10.10.10.2 qualify=600 disallow=all allow=ulaw canreinvite=no insecure=invite accountcode=pbx02 ;TESTS; ;[EMAIL PROTECTED] ;fromuser=pbx01topbx02 ;username=pbx01topbx02 ;secret=j48dj7rjd9023jd sendrpid=yes trustrpid=yes PBX02 [pbx01topbx02] type=friend host=10.10.10.3 qualify=600 context=dialOutPatternsAll disallow=all allow=ulaw canreinvite=no insecure=invite accountcode=pbx01 ;TESTS; ;[EMAIL PROTECTED] ;username=pbx01topbx02 ;fromuser=pbx01topbx02 ;secret=j48dj7rjd9023jd sendrpid=yes trustrpid=yes Layout SIPDEV -REGISTERD- PBX01 - PBX02 - PROVIDER With the above settings above I should be allowed to send calls as I like through the 2 boxes, but I can't I get the following messages: PBX01 -- Executing [EMAIL PROTECTED]:7] Dial(IAX2/4161231234-2, SIP/pbx01topbx02/16041231234) in new stack -- Called pbx01topbx02/16041231234 [Feb 1 11:35:18] NOTICE[13507]: chan_sip.c:11983 handle_response_invite: Failed to authenticate on INVITE to '4161231234 sip:[EMAIL PROTECTED];tag=as4a9e0ae9' -- SIP/pbx01topbx02-009c31d0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Then on PBX02 I don't get any debug/verbose messages unless I do a sip debug then I get: --- SIP read from 10.10.10.3:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK349431bd;rport From: 4161231234 sip:[EMAIL PROTECTED];tag=as5c5aac02 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: 4161231234 sip: [EMAIL PROTECTED];privacy=off;screen=yes Date: Fri, 01 Feb 2008 19:40:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 13479 13479 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 11026 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (15 headers 12 lines) --- Sending to 10.10.10.3 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] --- Reliably Transmitting (no NAT) to 10.10.10.3:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK349431bd;received=10.10.10.3;rport=5060 From: 4161231234 sip:[EMAIL PROTECTED];tag=as5c5aac02 To: sip:[EMAIL PROTECTED];tag=as19dfe789 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=6d7284c8 Content-Length: 0 Any insights as to why insecure invite is not taking effect? Bryan _ Bryan Cramer PO Box 616 Sechelt BC, V0N 3A0 Web: www.yellowions.com Tel: 604-773-4580 The information in this email is privileged and confidential. If you are not the intended recipient, please notify the sender immediately, as any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, or reference to it, is prohibited and potentially unlawful. ___ -- 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] LDAP support
There a realtime LDAP driver now in 1.6beta2 On 23/01/2008, Cavalera Claudio Luigi [EMAIL PROTECTED] wrote: Hello, I've found this information about asterisk and LDAP: http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP which can be out of date. I'm trying this http://www.mezzo.net/asterisk/app_ldap.html however I'm facing the same problems as this unanswered: http://forums.digium.com/viewtopic.php?p=42591sid=05e1d00ab6f9848f4e7b6 39d66cc6d79 Does anybody know how to solve this issue? Moreover I would like to understand if someone is using LDAP (for iax.conf) and with which asterisk plugin (e.g. app_ldap, Asterisk::LDAP Perl module, etc..). Best Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- 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 -- http://www.suretecsystems.com/services/openldap/ ___ -- 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] Enterprise or Fedora?
Matthew J. Roth wrote: love U.all wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference I started out running Fedora, but I have migrated away from it for a few reasons. Fedora has a very short life cycle and since the demise of the legacy project there is no support for releases that are over a year old. It is also considered a bleeding-edge distribution and is a testing ground for new applications and features. As such, it has frequent updates (including the kernel) that aren't guaranteed not to break your system. If stability and long-term support are your goals, I recommend taking a look at CentOS http://www.centos.org/. It is a binary-compatible clone of Red Hat Enterprise Linux that's free and has a very long support period. It's basically RHEL without the paid support contract. My migration to CentOS was painless, because the file system and configuration are practically identical to those of Fedora. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Milton Calnek BSc, A/Slt(Ret.) [EMAIL PROTECTED] 306-717-8737 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] It's about time! -- Digium PCI-Express Cards
Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Modulehttp://www.voipsupply.com/product_info.php?products_id=3352 It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? ___ -- 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] QueueMember event/LastCall Variable - Format?
Hi all, What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Thanks ! -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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] QueueMember event/LastCall Variable - Format?
On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I recall. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] It's about time! -- Digium PCI-Express Cards
Matt wrote: Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Module http://www.voipsupply.com/product_info.php?products_id=3352 It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? Digium already makes PCI Express analog cards - AEX800 and AEX2400. -- Jason Parker Digium ___ -- 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] SIP Softphones and Citrix ?
There is an option you might consider (if you are starting from scratch). Don't use citrix. Write a web app. Then embed a softphone in that web app. Tim. ___ -- 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's DANGEROUS Transfer CDR's
- Original Message From: Mindaugas Kezys [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 1 February, 2008 4:04:30 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's For such cases we usually suggest to put 2 boxes in your infrastructure: 1. Main billing gateway - where all PBX'es are connected (all client's remote PBX'es and your Local PBX) 2. Local PBX - where user's without PBX'es are connected Then user connects in following way: User - Local PBX - Main GTW - PSTN That way you will be save from transfer issue and all your clients will be able to transfer their calls on Local PBX. Hi Mindaugas, That's a good tip, thanks for that. My concerns would be that the call path is now running through two Asterisk servers and that could add some quality problems, probably negligible though. The other concern would be that for fault tolerance we'd now need double the number of servers. If we currently require 3 Asterisk load balanced servers then now we are going to need 6. It's an idea worth toying around with though. Maybe we could specify that all customers that required the ability to transfer had to use server x and then from the SIP Proxy only allow REFER requests to that server. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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] It's about time! -- Digium PCI-Express Cards
Jason.. great! How long have those analog cards been out? I don't see them on my suppliers list. Digium already makes PCI Express analog cards - AEX800 and AEX2400. -- Jason Parker Digium ___ -- 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] SIP Softphones and Citrix ?
I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of cisco softphones with Citrix will come into play with SIP softphones as well. Seems that the two biggest issues revolve around wrapping the UDP stream up with the ICA protocol, and possibly issues with the various mics and speakers and having to interface with them I think. However, I am also a firm believer that anything is possible, practical well not usually, and it may just be the time has not come yet for this. There is a good article about this over at: http://www.brianmadden.com/content/article/How-should-Citrix-integrate-VoIP-with-Presentation-Server Any thoughts, comments or insight into this and your experiences around any of this is appreciated. ___ -- 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] QueueMember event/LastCall Variable - Format?
Jared Smith wrote: On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote: What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I recall. Thanks. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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 1.4.17 and Teliax DTMF
At 10:57 AM 2/1/2008, you wrote: Any tweaks recommended for DTMF and Teliax? try: dtmfmode=auto That's what mine is now after rfc2833 stopped working. Ira ___ -- 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] Enterprise or Fedora?
shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- 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] It's about time! -- Digium PCI-Express Cards
On Friday 01 February 2008 15:31, Matt wrote: It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? I have been using 220B's for about 6 months. I have about 20 of them out in the field. I have not had any issues with them, and feedback is positive. Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to make SIP calls through Asterisk with anonymous connection
I am trying to setup SIP to SIP calling between Asterisk managed networks. I want to make it so that people can call SIP:[EMAIL PROTECTED] and they connect to my Asterisk and get my external IVR then they can dial my extension or navigate extensions just like they would if they had called using a PSTN line. I also want to call other people using my Asterisk and dialing an external SIP like so SIP:[EMAIL PROTECTED] I want out going SIP calls to me managed by my Asterisk so I can transfer them to other people in my office or conference or use any of the other great features that Asterisk provides. I do not want to go SIP direct to SIP, I want to go SIP to Asterisk to Asterisk to SIP and connect to the far end Asterisk without requiring me to register my Asterisk server with the far end Asterisk server. For testing I have setup two servers running Asterisk. Both are on the Internet with static IP addresses and behind firewalls. The firewalls are configured to allow TCP UDP ports 5060 to 5082 and 10001 to 2 to connect directly to the Asterisk servers. This allows SIP and RTP connections from the outside. I have tested with Twinkle (a Linux softphone) and can connect to a registered account with NAT from external IPs. I have also set the Asterisk servers to allow incoming anonymous SIP calls to connect to the from-external. When I try to dial SIP:[EMAIL PROTECTED] Asterisk tries to dial the some_extension on my local network not the other network. I reconnect to the running asterisk using -r and watch when I dial and it does not report the @other_url only the some_extension. I am not having much luck finding the documentation I need. Can someone point me to a How-To on doing this? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- 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] X-Lite Softphone keeps de-registering?
The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome! ___ -- 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] Real API for Perl?
Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI binding improved since then? 2) Is there any chance of a real API for Perl? Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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 1.4.17 and Teliax DTMF
Carlos Chavez wrote: I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because DTMF is not understood. If I dial from a land line I can get in with no problems. Any tweaks recommended for DTMF and Teliax? I've had no issues with our Teliax accounts since switching to 1.4.x. I stayed back on 1.4.16.2 so far because of one issue with parking calls. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] IAX Registraion Refresh
I have Asterisk 1.4 registering via IAX to another Asterisk machine. How can I change the default registration timeout of 60s? I need my Asterisk box to register every HOUR Anyone? Editting source isn't an option. Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] Echo() app doesn't work
Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details: A. Platforms: -- AsteriskNOW 0.6 beta 32bit, updated; -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2 -- Debian Etch 32bit with custom compiled kernel (Timer frequency 1000 Hz and couple more tweaks) and latest stable asterisk (1.4.17) compiled from source -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10) -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10) Echo() works only on the 64-bit setup. Does not work for all other cases. The Playback() app works fine in *all* cases. (The microphone is tested and works fine, so it's not that simple!) For some of the setups I established two separate extensions and they could talk to each other (so important things work, yes). The logs show the same, that is, just what would be normal: -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- My extentions.conf: -cut here--- [globals] [general] [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten = 500,1,Verbose(1|Echo test application) exten = 500,n,Echo() exten = 500,n,Hangup() exten = 501,1,Verbose(1|Playback test application) exten = 501,n,Playback(vm-review) exten = 501,n,Wait(1) exten = 501,n,Hangup() [phones] include = internal -cut here--- My iax.conf: -cut here--- [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no autokill=yes [yassen] type=friend host=dynamic context=phones -cut here--- Anyone having a suggestion what might be the reason for the nonworking Echo() ? I am really stuck; Google could not really help. Any ideas would be highly appreciated! Thanks in advance, Yassen -- Yassen Damyanov Adelie Ltd. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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] Polycom - Buddy Watch not a choice when adding Speed Dial
Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at Auto Divert.I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add bw1/bw manually to the speed dial file it doesn't change anything. The buttons work well for a speed dial. The icon next the speed dial is 10 dots, in the shape of a keypad. Anyone else experience this? Thanks, Thermal ___ -- 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] Polycom - Buddy Watch not a choice when adding Speed Dial
On Feb 1, 2008 3:53 PM, Thermal Wetland [EMAIL PROTECTED] wrote: Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at Auto Divert.I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add bw1/bw manually to the speed dial file it doesn't change anything. The buttons work well for a speed dial. The icon next the speed dial is 10 dots, in the shape of a keypad. Anyone else experience this? Thanks, Thermal Check your sip.cfg for the line: feature.1.name=presence feature.1.enabled=1 I would imagine that you have enabled=0 -Matt ___ -- 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] Echo() app doesn't work
On Fri, Feb 01, 2008 at 05:01:56PM -0800, Yassen Damyanov wrote: Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details: A. Platforms: -- AsteriskNOW 0.6 beta 32bit, updated; -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2 -- Debian Etch 32bit with custom compiled kernel (Timer frequency 1000 Hz and couple more tweaks) and latest stable asterisk (1.4.17) compiled from source -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10) -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10) Echo() works only on the 64-bit setup. Does not work for all other cases. The Playback() app works fine in *all* cases. (The microphone is tested and works fine, so it's not that simple!) For some of the setups I established two separate extensions and they could talk to each other (so important things work, yes). The logs show the same, that is, just what would be normal: -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- On which platform is that? Echo is executed, and exists without an error. My extentions.conf: -cut here--- [globals] [general] [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten = 500,1,Verbose(1|Echo test application) exten = 500,n,Echo() exten = 500,n,Hangup() exten = 501,1,Verbose(1|Playback test application) exten = 501,n,Playback(vm-review) exten = 501,n,Wait(1) exten = 501,n,Hangup() [phones] include = internal -cut here--- My iax.conf: -cut here--- [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no autokill=yes [yassen] type=friend host=dynamic context=phones -cut here--- Anyone having a suggestion what might be the reason for the nonworking Echo() ? I am really stuck; Google could not really help. Any ideas would be highly appreciated! Thanks in advance, Yassen -- Yassen Damyanov Adelie Ltd. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI card with PCI-E interface
Hi Alberto, 2008/2/1, Alberto Pastore [EMAIL PROTECTED]: Olivier ha scritto: Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The Sangoma A500BRX is a 2 to 6 bri pci-x interface, although I've never tested it (A500BRECX comes with hw echo cancellation). Could you slide it in 1U rack server. If my memory serves me right, Sangama cards must sometime be stacked specifically. Regards, Alberto. -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ -- 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] Real API for Perl?
Ken D'Ambrosio wrote: Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI binding improved since then? 2) Is there any chance of a real API for Perl? What is your criterion of real? That is to say, what do you need that it does not provide? I've used AGI and FastAGI in Perl extensively and it is yet to fail to serve my purposes. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] X-Lite Softphone keeps de-registering?
Doug wrote: The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome! I suppose the usual answers apply. You can have Asterisk bind to UDP port 80 in addition to port 5060, or you can do destination NAT. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
