Re: [Asterisk-Users] FritzCard, mISDN Anlagenanschluss
Yes, it is possible. I'm using PtP and TE mode at home with chan_misdn. Hans Ralf Mueller schrieb: Hello, can someone on the list confirm, that it is possible to connect a FritzCard to an Anlagenschluss, when I use the mISDN driver? I have read a number of posting and articles, that this is not possible with the CAPI driver, but found no clear answer about the mISDN driver. Thanks for your help, Ralf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1/3 packets are reported dropped by tethereal
HiWhen i ran the below command on vicidial dialer:[EMAIL PROTECTED] ~]# tethereal -i eth0 -a duration:300 -w sample.capCapturing on eth0320167147496 packets droppedon net i found: When i ran Acterna PVA-1000 on sample.cap it showed Max Jitter about 20 % and packet loss and echo as major cause of voice degradation. MQS was also less 2.59 where as it should be around 5.0. are packets being dropped at interface card or at kernel and how to correct it. Machine configuration is given below: So 1000 packets captured means, in Tethereal and tcpdump, 1000 packets read from libpcap and processed. What 100 packets dropped means is that, of all the packets supplied to the kernel's packet capture mechanism that passed the filter, 100 of them were dropped because there wasn't enough room in the kernel's buffer for them; this means that packets aren't being read from the kernel's capture mechanism fast enough by the application.Machine configuration:Linux vicidial2.esselshyam.net 2.6.11-1.1369_FC4smp #1 SMP Thu Jun 2 23:08:39 EDT 2005 i686 i686 i386 GNU/Linux Machine Model: HP dx6120 MTRAM 1 GBHDD 80 GB SATA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sending special infoa fter login
hello all Isit possible to send special informations to a phone after it registered? i want to send some config infos to the phone after it registered to the *. Is that possible? And if yes how? regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
2006/4/23, Armin Schindler [EMAIL PROTECTED]: On Sun, 23 Apr 2006, Olivier Krief wrote: 2006/4/21, Armin Schindler [EMAIL PROTECTED]:But if you want to forward a call (which was already accepted by Asterisk) to another CAPI application, it is not possible. (Well, Eicon has a special driver which can do a lot of CAPI extensions, but I did not try this yet). So if you want to do that, I suggest using just chan-capi for receiving faxes and maybe another application for sending faxes. Armin This is exactly the heart of my question : is it possible to accept a call with capi-enabled Asterisk, detect it's a fax and forward it somehow to a hardware-enforced fax application on the same server. Doing that you could get higher fax speeds and reliability and interesting Asterisk features.Why do you think you will get higher fax speeds or reliability? There is no difference in receiving a fax over CAPI between chan-capiand any fax-software like capi4hylafax. Both use the same CAPI commandswith the use of the card's fax capabilities.I thought that receiving fax with Asterisk always meant to use one way or another, spandsp library. That's the reason why I thought fax-enabled boards provide higher fax speeds or reliability.(I don't mean using spandsp isn't reliable : I mean fax boards are said and priced to be more reliable and it's worth to evaluate the benefit of using them). When writing receiving a fax over CAPI, do you mean receiving a fax over CAPI with Asterisk and processing it with spandsp ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Olivier Krief wrote: When writing receiving a fax over CAPI, do you mean receiving a fax over CAPI with Asterisk and processing it with spandsp ? No, with a full Eicon Diva 4BRI card, it does hardware faxing. Instead of using rxfax (which uses spandsp), you'd use capicommand(receivefax) which does a hardware receive on-board. Also, I can confirm that you can receive faxes using spandsp on the V-4BRI (voice-only) board. Which is nifty. :) cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] annoying noise on analog phones on tdm400p
Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? cheers, tom ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Mon, 24 Apr 2006, Olivier Krief wrote: 2006/4/23, Armin Schindler [EMAIL PROTECTED]: On Sun, 23 Apr 2006, Olivier Krief wrote: 2006/4/21, Armin Schindler [EMAIL PROTECTED]: But if you want to forward a call (which was already accepted by Asterisk) to another CAPI application, it is not possible. (Well, Eicon has a special driver which can do a lot of CAPI extensions, but I did not try this yet). So if you want to do that, I suggest using just chan-capi for receiving faxes and maybe another application for sending faxes. Armin This is exactly the heart of my question : is it possible to accept a call with capi-enabled Asterisk, detect it's a fax and forward it somehow to a hardware-enforced fax application on the same server. Doing that you could get higher fax speeds and reliability and interesting Asterisk features. Why do you think you will get higher fax speeds or reliability? There is no difference in receiving a fax over CAPI between chan-capi and any fax-software like capi4hylafax. Both use the same CAPI commands with the use of the card's fax capabilities. I thought that receiving fax with Asterisk always meant to use one way or another, spandsp library. That's the reason why I thought fax-enabled boards provide higher fax speeds or reliability. (I don't mean using spandsp isn't reliable : I mean fax boards are said and priced to be more reliable and it's worth to evaluate the benefit of using them). When writing receiving a fax over CAPI, do you mean receiving a fax over CAPI with Asterisk and processing it with spandsp ? When using a card with onboard DSPs (or even the software fax of AVM Fritz binary-only driver) you can do faxing with the CAPI interface. That means you don't get the audio data stream, you get the fax-data instead which can be save in a file. In that case the application (Asterisk or anything else) don't need to do the fax processing, this is done by the driver (in case of AVM Fritz) or on the hardware in case of DSPs (like Eicon DIVA Server). chan-capi supports this and if the CAPI driver/device supports fax over CAPI, you don't need anything like spandsp. That's what I mean with faxing over CAPI. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Polarity Reverse Detection
Hi, I have read many postings but still can not understand - is it possible the X100P to detect a polarity reverse, when the call is answered and when it ends? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback(something,noanswer) on Zap?
On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote:
A better solution is to set the PRI hangup cause before dropping the
incoming call; if you set the hangup cause to 'number not assigned'
then your telco's switch will play its normal intercept message to
the caller.
Thank you! This works!
context from-e1 {
_X. = {
AGI(pub2ext.agi);
PRI_CAUSE=1;
Hangup();
};
};
Now caller hears voice from his/her telco (not from my telco) saying
that number is not available. This is even better.
___
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[Asterisk-Users] MeetMe Call Out to invite
hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Hi Armin! Armin Schindler wrote: On Fri, 21 Apr 2006, Klaus Darilion wrote: Hi! I've forgotten to ask an important question: Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode? Yes, and each port can be configured separately. I guess also NT/TE can be configured for each port separately? What about manually disabling the onboard echo canceler for certain extensions. Can this be done with a certain capicommand()? Out of curiosity: Are there also software echo cancelers available (if for some reason I do not like the HW canceler) like in chan_zap? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe Call Out to invite
You can just call them from your dialplan and make them join a meetme room.If by application you mean a frontend, you can use web meetme (juste search for web meetme) to invite new participant from a web browser. b.en.qOn 4/24/06, welemon lee [EMAIL PROTECTED] wrote: hi all, is there a kind of application canlet asterisk call outfellows, and invite them to come to join the meetme.these fellows do not need to call in asterisk , just wait for a call. 3xwelemon___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Mon, 24 Apr 2006, Klaus Darilion wrote: Hi Armin! Armin Schindler wrote: On Fri, 21 Apr 2006, Klaus Darilion wrote: Hi! I've forgotten to ask an important question: Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode? Yes, and each port can be configured separately. I guess also NT/TE can be configured for each port separately? Yes, you can even run different ISDN D-channel protocols. What about manually disabling the onboard echo canceler for certain extensions. Can this be done with a certain capicommand()? Not yet, but this will be available in the next version of chan-capi. Currently it is on a per port basis only. Out of curiosity: Are there also software echo cancelers available (if for some reason I do not like the HW canceler) like in chan_zap? chan-capi still has the old echo-squelch, but this is a very primitive one. Besides that, there is no software-echo-cancel in CAPI (chan-capi). I think such common software features, as well as jitterbuffer, does not belong into a channel module, it is core functionality. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. I have just returned from one week in Tokyo, testing the Asterisk SIP channel with a large number of other SIP stacks. Since last SIPit, the SIP stack has improved quite a lot. It looks really promising for the 1.4 release of Asterisk this summer. Things are progressing well. At the next SIPit in New Hampshire this fall, I hope to have the first version of chan_sip3 for testing. Asterisk is a world wide project with many members. In Tokyo, I met Japanese Asterisk users and learned quite a lot on how they use Asterisk. An Open Source project is not only about software, it's also about the people involved. You are all very important, your feedback and support are our keys to success. Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel, the bug tracker or on the mailing list! /oej ** Asterisk European Tour - MeetAsterisk.com! This week and next week there's a European tour with Asterisk seminars for beginners, named MeetAsterisk. The event is organized by Edvina.net in cooperation with Digium, Xorcom and local Asterisk distributors and consultants. Register now to make sure you have a seat! - http://www.meetasterisk.com ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the release version. The release version is distributed as .tar.gz archives on several servers. The current released version of Asterisk is 1.2.7.1. The release version is fixed, we are adding no new functions and only changes it when bugs are fixed. Current versions: - Asterisk Version 1.2.7.1 - Zaptel Version 1.2.5 - Libpri Version 1.2.2 - Addons Version 1.2.2 - Sounds Version 1.2.1 The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. The development version is branded 1.3 and will be the basis for the next release version, version 1.4. There are also a lot of development branches in our subversion repository, hosting new functionality developed for testing by you, the Asterisk community. For more information about these, please visit http://www.voip-forum.com/index.php?p=189more=1 ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. And please do not send out test messages to the list. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org You can download their new book from the web site or buy it from the bookstore. * Asterisk Daily news is at http://www.sineapps.com/news.php * VoIP-search (Asterisk mailing list etc) http://search.voip-forum.com Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. Do not use this list as a secondary support line if you do not get an answer on the -users list. It is meant for developer discussions, not advanced support. If you need answers, there is a better chance that you will get help on
[Asterisk-Users] Re: MeetMe Call Out to invite
In article [EMAIL PROTECTED], welemon lee [EMAIL PROTECTED] wrote: hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. You could try adapting the patch from http://bugs.digium.com/view.php?id=3405 It's quite old, so you will most likely have to apply it by hand. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe Call Out to invite
1. u need to schedule the call -- u can do it with something like this: http://www.voip-info.org/wiki/view/Asterisk+tips+wake-up 2. just call all the participants. check out the GOTO or G in Dial() application. it will send the called peer to an extension u want. u just need to make them join the meetme room at that extension. welemon lee wrote: hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pimjai Wesnarat nummerndirekt GmbH Oskar-Jäger-Str. 125 50825 Köln Tel.: +49 (0)221 2601571 Fax : +49 (0)221 2601579 http://www.nummerndirekt.de ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: SPA 3000 - UK Replacement
After my ongoing experience of being a customer of Currys (part of Dixons Group PLC), I can only suggest that NOONE SANE should ever purchase anything from any part of Dixons Group PLC, including Currys, Dixons PCWorld MasterCare. They are a total waste of time/money. Liars, Corporate liars, and Damn Corporate Liars. Bails 0 Steven wrote: First off I am totally annoyed and let down by PC World Business (PCWB part of the Dixons Group). I ordered one of these babies from them over a month ago. After constantly chasing them up they finally told me they couldn't deliver, and have now only just returned the money they stole from me. I only bought from them because they showed a 4-day availability stock level! Now I'm screwed as it seems these are impossible to come by in the UK now since Sipura decided to discontinue it.. Now my back to my subject. Does anyone know of a decent replacement for the SPA3000. I need at least 1 FXO and 1 FXS but am willing to pay for 2 FXS on the same unit. What I'm looking for needs to be of a likable price to the SPA3000 which in it's hayday was retailing for around £70 at some outlets. I'm primarily looking for something network attachable. But could stretch to USB or PCI if the price was right.. I'm steering away from PCI cards as they seem to have terrible issues with UK analogue lines such as not being able to detect hang ups.. (Also; the server I'd ideally like to add this capability too, has no free PCI slots..) Thanks for your time, Steve Daniels (I've been trying to send this for ages but crappy outlook keep sending it from another of my email accounts despite the fact that I keep setting it to send it via the correct one!) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: None To: [EMAIL PROTECTED] Subject: Microsoft Office Outlook Test Message This is an e-mail message sent automatically by Microsoft Office Outlook's Account Manager while testing the settings for your POP3 account . ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Hi all, Where can we find a roadmap of asterisk 1.4 release ? Harry --- Olle E Johansson [EMAIL PROTECTED] a écrit : Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. I have just returned from one week in Tokyo, testing the Asterisk SIP channel with a large number of other SIP stacks. Since last SIPit, the SIP stack has improved quite a lot. It looks really promising for the 1.4 release of Asterisk this summer. Things are progressing well. At the next SIPit in New Hampshire this fall, I hope to have the first version of chan_sip3 for testing. Asterisk is a world wide project with many members. In Tokyo, I met Japanese Asterisk users and learned quite a lot on how they use Asterisk. An Open Source project is not only about software, it's also about the people involved. You are all very important, your feedback and support are our keys to success. Again, welcome to the Asterisk.org Open Source PBX Project! Meet you on the IRC channel, the bug tracker or on the mailing list! /oej ** Asterisk European Tour - MeetAsterisk.com! This week and next week there's a European tour with Asterisk seminars for beginners, named MeetAsterisk. The event is organized by Edvina.net in cooperation with Digium, Xorcom and local Asterisk distributors and consultants. Register now to make sure you have a seat! - http://www.meetasterisk.com ** Asterisk version information At this moment we have two current versions of Asterisk, the developer version and the release version. The release version is distributed as .tar.gz archives on several servers. The current released version of Asterisk is 1.2.7.1. The release version is fixed, we are adding no new functions and only changes it when bugs are fixed. Current versions: - Asterisk Version 1.2.7.1 - Zaptel Version 1.2.5 - Libpri Version 1.2.2 - Addons Version 1.2.2 - Sounds Version 1.2.1 The development version is to be used by people that can test new functions and live with bugs and unexpected shortcomings. The development version is branded 1.3 and will be the basis for the next release version, version 1.4. There are also a lot of development branches in our subversion repository, hosting new functionality developed for testing by you, the Asterisk community. For more information about these, please visit http://www.voip-forum.com/index.php?p=189more=1 ** The mailing list is growing Today, we propably have over 10,000 readers on the -users list. This means that everything anyone write to this mailing list, is sent to thousands of mailboxes that are already flowing over with messages. That's why we all need to follow some simple rules on how to use the mailing list and the other tools that are available. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. And please do not send out test messages to the list. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org You can download their new book from the web site or buy it from the bookstore. * Asterisk Daily news is at http://www.sineapps.com/news.php * VoIP-search (Asterisk mailing list etc) http://search.voip-forum.com Finally, if you don't find the answer elsewhere, try the list. **
[Asterisk-Users] Dreadful results from zttest with TE210P and Dell 2850?
Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling zaptel-1.2.5
Hello, What's wrong ? make install . options torisa base=0xd alias char-major-196 torisa alias wcfxs wctdm alias wct2xxp wct4xxp if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/ztd-eth.o [ `id -u` = 0 ] /sbin/depmod -a 2.4.27-2-386 || : depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/ztd-eth.o [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf == serveur1:/usr/local/src/ASTERISK/zaptel-1.2.5# modprobe zaptel /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rf7663209 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol add_wait_queue_R2cea9688 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R31ed257b /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_wait_queue_Ree3648ba /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol __pollwait_R35631129 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol create_proc_entry_R648035a2 /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod /lib/modules/2.4.27-2-386/misc/zaptel.o failed /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod zaptel failed Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?
I think it is correct. Isn't that why they call it a Smart Jack? I've only ever seen a regular cat5 cable used from the Smart Jack to the device (router/PBX/CSU/DSU/whatever). I believe the point of the smart jack is, amongst other things, to allow for the use of readily available cables. I agree however that back-to back (PBX-PBX etc) you would need a cross-over cable. Mark On Sat, 2006-04-22 at 18:14 -0400, Steven Totaro wrote: The telco guys probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ? att (formerly SBC, formerly Southwestern Bell, formerly ATT) just came out and installed my PRI. FYI, they used Cat 5e cable. No special T1 cabling that costs a fortune to buy somewhere, just plain old Cat 5e cable. Guess what guys? If they are using this as customers' sites, they are probably using it elsewhere. It's only as good as the weakest link, so you can go out and spend lots of money on T1 cable, or just use Cat 5e like the telco guys do. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk @ Home install on server with Apache already running
I've just completed downloading and installing [EMAIL PROTECTED] on my already running server, and came to view the config areas, however I already have Apache installed, and operating on the server, so viewing port 80 takes me straight to the relevant apache served page. Is there a way to change the port on which [EMAIL PROTECTED] serves? Cheers Nunners ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hi...Please help me
Hi Friends,I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is:PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic)--- Employee 2 PC (Softphone i.e., Headphones with Mic)--- Employee 3 PC (Softphone i.e., Headphones with Mic) --- -- --- ----- Employee 10 PC (Softphone i.e., Headphones with Mic)and vice versa.How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank youThanks Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1/min.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error messages
Are there some instructions how to solve problems that produce some typical error messages in asterisk? For example, if I don't use iax, dundi or mysql logging, every time I start asterisk I'll get several error messages. How What can I do to disable loading those files? Here re some error messages that I receive. Apr 24 12:21:09 ERROR[2050] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Apr 24 12:21:09 WARNING[2050] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Apr 24 12:21:09 WARNING[2050] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Apr 24 12:21:09 WARNING[2050] pbx.c: Requested contexts didn't get merged Apr 24 12:21:10 ERROR[2050] pbx_dundi.c: Unable to load config dundi.conf Apr 24 12:21:10 ERROR[2050] chan_iax2.c: Unable to load config iax.conf Apr 24 12:21:12 WARNING[2050] cdr_custom.c: Failed to load configuration file. Module not activated. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.2.4/7 and chan_modem
Hi, I am currently running several * boxes on 1.0.9 with HFC chipset ISDN modems using i4l's hisax driver and chan_modem. Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7 or will I need to change it to use bristuff or chan_capi? I want to do the upgrade with as little changes as possible. Thank you Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement System for a Charity
Yes its possible, just create different contexts for each organisation. Bails Michiel van Baak wrote: On 13:40, Thu 20 Apr 06, Douglas Garstang wrote: Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. No, that is not possible with AMP/freepbx One of the reasons why I trashed it :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound calls to sip urls
Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. Regards, Ajit ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2.4/7 and chan_modem
Marnus van Niekerk wrote: Hi, I am currently running several * boxes on 1.0.9 with HFC chipset ISDN modems using i4l's hisax driver and chan_modem. Will I be able to use my existing chan_modem setup with 1.2.4 or 1.2.7 or will I need to change it to use bristuff or chan_capi? I want to do the upgrade with as little changes as possible. Thank you Marnus van Niekerk -- chan_modem is still shipped with 1.2.7, you just need to uncomment line 21 in the Makefile that is in the channels folder of the source before compiling asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk @ Home install on server with Apache already running
I just started another html process running as user asterisk and group asterisk and changed the listen directive to another port. I am not running asterisk at home, but freepbx which is very similar, so I am told. on Monday 04/24/2006 James Nunnerley([EMAIL PROTECTED]) wrote I've just completed downloading and installing [EMAIL PROTECTED] on my already running server, and came to view the config areas, however I already have Apache installed, and operating on the server, so viewing port 80 takes me straight to the relevant apache served page. Is there a way to change the port on which [EMAIL PROTECTED] serves? Cheers Nunners ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Can't anyone stop self-promotion and tell the poor guy what he needs. A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows: 1 - 4 2 - 5 3 - NU 4 - 1 5 - 2 6 - NU 7 - NU 8 - NU NU = Not Used I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. It would probably be helpful for everyone reading this thread to understand what the differences are in the two types of cables. Primarily impedance matching, twists per foot, shielding, etc. For short runs, the use of cat5 vs proper T1 cables isn't likely to have any impact unless there is a fair amount of induction from electrical noise, etc. That can take the form of florescent fixtures, transformers, older CRT monitors, etc, etc. On longer runs, the shielded T1 cables are likely to provide better results particularly if there happens to be any electrical noise. Oh, and if shielded T1 cable is used, the shield at each end of the cable must be grounded. (Let's see how many can figure out how to do that via an rj45 plug. ;) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum D3000
Please has anyone on this list had experience with getting Quintum equipment to talk to Asterisk? Specifically a D3000 in my case. It is refusing to register and I'm out of ideas. Any help appreciated. Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] compiling zaptel-1.2.5 [SOLVED]
--- [EMAIL PROTECTED] a écrit : Hello, What's wrong ? make install . options torisa base=0xd alias char-major-196 torisa alias wcfxs wctdm alias wct2xxp wct4xxp if [ -d /etc/modutils ]; then \ /sbin/update-modules ; \ fi depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/ztd-eth.o [ `id -u` = 0 ] /sbin/depmod -a 2.4.27-2-386 || : depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/wctdm24xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/ztd-eth.o [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf == serveur1:/usr/local/src/ASTERISK/zaptel-1.2.5# modprobe zaptel /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rf7663209 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol add_wait_queue_R2cea9688 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R31ed257b /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol remove_wait_queue_Ree3648ba /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol __pollwait_R35631129 /lib/modules/2.4.27-2-386/misc/zaptel.o: /lib/modules/2.4.27-2-386/misc/zaptel.o: unresolved symbol create_proc_entry_R648035a2 /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod /lib/modules/2.4.27-2-386/misc/zaptel.o failed /lib/modules/2.4.27-2-386/misc/zaptel.o: insmod zaptel failed Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendezvous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPredirect [2]
--- [EMAIL PROTECTED] a écrit : Hello, I read http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPredirect I wish to configure asterisk as a redirect server. I have badly understood this command . ASTERISK | sip agents nated ==SER When sip agents send INVITE to the proxy i want ser send this one to ASTERISK . How asterisk can send 302 messages with the good info in the sip contact HF ? Can asterisk get the sip uri via dns srv lookup ? Does asterisk look up in its own location ? Thanks for examples Regards Harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendezvous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting up a t38 fax gateway [2]
--- [EMAIL PROTECTED] a écrit : Hello to all, Is there an how-to for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendezvous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User Defined VoiceMail announcement?
Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpINWXK4ofcq.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error messages
Tomislav Parčina wrote: Are there some instructions how to solve problems that produce some typical error messages in asterisk? For example, if I don't use iax, dundi or mysql logging, every time I start asterisk I'll get several error messages. How What can I do to disable loading those files? Here re some error messages that I receive. Apr 24 12:21:09 ERROR[2050] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Apr 24 12:21:09 WARNING[2050] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Apr 24 12:21:09 WARNING[2050] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Apr 24 12:21:09 WARNING[2050] pbx.c: Requested contexts didn't get merged Apr 24 12:21:10 ERROR[2050] pbx_dundi.c: Unable to load config dundi.conf Apr 24 12:21:10 ERROR[2050] chan_iax2.c: Unable to load config iax.conf Apr 24 12:21:12 WARNING[2050] cdr_custom.c: Failed to load configuration file. Module not activated. -- well, if you dont use/need a module, in modules.conf put noload = app_intercom.so (for example). i think you can choose whether to automatically load all then specifically noload whichever you dont want with a noload =, or with autoload=no, specify which you want to load. -- thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Rich Adamson wrote: snip For short runs, the use of cat5 vs proper T1 cables isn't likely to have any impact unless there is a fair amount of induction from electrical noise, etc. That can take the form of florescent fixtures, transformers, older CRT monitors, etc, etc. On longer runs, the shielded T1 cables are likely to provide better results particularly if there happens to be any electrical noise. Oh, and if shielded T1 cable is used, the shield at each end of the cable must be grounded. (Let's see how many can figure out how to do that via an rj45 plug. ;) Totally agree with you, unshielded cables are only usable if the distance is short. Just curious, how should one ground the shield? Do you ground it to the ground bar in the server room? Any special requirements? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] answer delay
Hi Folks,using this:exten = x,1,Playback(audio,noanswer)exten = x,2,Answerexten = x,3,BackGround(out)exten = x,103,HangupI'm not billed and remain connected, but the file 'audio' is not played...well I do not ear it. But after, it pass correctly to answer and I can ear the 'out' audio file.Any idea/suggestion???Thanks!2006/3/21, FaberK [EMAIL PROTECTED] :Hi,I've tryed it using my mobile and I've been charged. Maybe, my mobile operator(Vodafone) does not support it?Thanks again.p.s.: hi John, I love to learn(books, google, lists, ecc...) and cooperation and I can say that everytime I learn something new. :o) 2006/3/21, CC Jay [EMAIL PROTECTED]: Not sure about 1.2.4 but with 1.0.9, you'll need to add noanswer to playback since playback will try to answer the line, i.e.,exten = 5551234,1,Playback(you-wont-be-billed-for-hearing-this, noanswer) exten = 5551234,n,Answeretc. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Leo Ann Boon wrote: Rich Adamson wrote: snip For short runs, the use of cat5 vs proper T1 cables isn't likely to have any impact unless there is a fair amount of induction from electrical noise, etc. That can take the form of florescent fixtures, transformers, older CRT monitors, etc, etc. On longer runs, the shielded T1 cables are likely to provide better results particularly if there happens to be any electrical noise. Oh, and if shielded T1 cable is used, the shield at each end of the cable must be grounded. (Let's see how many can figure out how to do that via an rj45 plug. ;) Totally agree with you, unshielded cables are only usable if the distance is short. Just curious, how should one ground the shield? Do you ground it to the ground bar in the server room? Any special requirements? That would be one way to do it, at both ends. Personally, I would not try to install an rj45 on each end of a shielded T1 cable, but rather terminate the cable on a patch panel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. He told me that with the number of twists in the Cat 5 cable it would cancel out any interference, but he also stated that the effective length was calculated using a cable with less twists and subsequently 'less dense' and that if using a Cat5e cable you must factor that in. so if you use cat5e cable your are fine but you can't go as far. Regarding the Smart Jack it is mostly used as a location at the CPE where the Telco can loop and make sure that the problem is at your end. So your assumption is correct that you can plug anything you want into it, its one your side of the demark, so if it doesn't work it's YOUR problem. Totally agree with you, unshielded cables are only usable if the distance is short. Just curious, how should one ground the shield? Do you ground it to the ground bar in the server room? Any special requirements? That would be one way to do it, at both ends. Personally, I would not try to install an rj45 on each end of a shielded T1 cable, but rather terminate the cable on a patch panel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium cards for sale
Hi, We got following surplus for sale: TE210P $700 TE410P $1100 TE411P $1950 Bundle (All 3 cards) please make an offer :) Cards not used except for development testing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue reload
I've noticed that when app_queue.so is reloaded(or just a reload command is used) that all queue members that were paused are automatically unpaused. Is there a workaround for this? (Note, I use statically defined callback agents). --johann ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange problem with Telasip DID, please help
I have configured telasip DID with following entried in sip_custom.conf [telasip] username= (fake) type=peer secret=x quality=yes nat=yes insecure=very fromuser= host=gw4.telasip.com #disallow=all #allow=ulaw #allow=alaw fromdomain=gw4.telasip.com context=from-telasip and Register string in sip.conf under general and extensions.conf has following entries [from-telasip] exten = 1134817097,1,Answer exten = 1134817097,2,Wait(1) exten = 1134817097,3,Background(pls-hold-while-try) exten = 1134817097,4,NoOp(Incoming call for Suzie on TelaSIP #8431234567) exten = 1134817097,5,Dial(SIP/71469,20,m) exten = 1134817097,6,VoiceMail([EMAIL PROTECTED]) exten = 1134817097,7,Hangup The problem is I can receive one incoming call to this DID successfully. Then I tried to call this DID, it say it is not avaiable. SO in Asterisk CLI I type reload to reload Asterisk. Then incoming call works again, then next one is not, then reload, it works, so and so. What could be the problem? Please help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.
Thank you.
It SEEMS to be working fine now as-is with the cranked-up registration time.
When the time comes to tinker with it in the future - I will probably try
working with groups again, or even work something out with astdb. (and,
most likely, end up breaking something that seems to work already)
I did have something similar to the problem outlined in the chain: RE:
[Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strangeproblem
Where:
- Call is in progress from USER1 - Asterisk - SPA - PSTN_Person
- SPA indicates an incoming call (existing call is still in progress).
Asterisk rings phones, USER2 answers
- USER2 answers, and ends up talking to PSTN_Person.USER1 is
disconnected.
I was playing with the spa3k's settings at the time, and attributed that
instance to my actions.
Another instance:
- Call is in progress USER1 - Asterisk - SPA - PSTN_Person
- SPA indicates an incoming call (existing call is still in progress).
Asterisk rings phones, USER2 (me - in this case) answers
- USER1 and USER2 and PSTN_Person end up in a 3-way call.
- USER2 hangs up,PSTN_Person is disconnected.
THIS occurred very close to the same time as the unit re-registering (I made
some configuration changes and reset the box an hour earlier, with the
registration time set to expire at 3600 seconds), and is what started me
looking at it.
My test of making it hammer on the registrations isn't really a fair
comparison to production use, and doesn't help in reproducing the two
scenarios above - but it does seem to indicate that there is an issue in the
registering code. (A dropped call is reproducible on 2 of my spa3k's -
haven't tested the other two).
I guess what I am suggesting: as part of diagnosing an erratic behaviour
problem with an spa3k, look at the registering time(s). I would be
really interested if there is a correlation.
- Original Message -
From: Moises Silva [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 20, 2006 7:34 AM
Subject: Re: [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.
i just got a SPA3000 but still not using it on production, and i
havent tested deeply. However, have you tried using incominglimit=1
in the register context of the SPA?? i guess that would limit in the
PBX rather that sending the call to the SPA.
Regards
On 4/20/06, Dana Harding [EMAIL PROTECTED] wrote:
Hello All!
I am in the process of assembling an asterisk-based phone system for my
office - using SPA-3000s to connect the network to the PSTN. I am
wondering if anybody else can get (or has already seen) the same behaviour
out of their 3000.
The short version: Send multiple Calls to the SPA's FXO port at the same
time it is re-registering with Asterisk.
SPA HTTP Configuration: PSTN Line - Register Expires: 5
(to ensure it is registering all the time)
Dial one number through the SPA's FXO port - establish a conversation
Dial another number through the same FXO port (SPA3000/NXY).
What SHOULD happen is the second caller receives a '504 - Service
Unavailable' error while the first caller happily continues the
established
conversation. What happens here: the already established call gets
dropped, AND the second caller gets a 504 error.
I did send a note to Linksys - and will see what kind of reponse they
have.
With longer Register Expires: times (10, 30, 60 seconds) it took more
attempts to make the call drop.
I have my Register Expires time cranked up to 86400 (1 day) now - and am
hoping I don't see another repeat.
---
There are three SPA-3000s in the system. I looked at some more
complicated dialplan layouts, and decided to keep it simple:
exten = s,1,Dial(${PSTN2}/${ARG1},,n)
exten = s,2,Dial(${PSTN3}/${ARG1},,n)
exten = s,3,Dial(${PSTN1}/${ARG1},,n)
exten = s,4,Wait(1)
exten = s,5,Playback(all-circuits-busy-now)
exten = s,6,Congestion()
PSTN1,2,3 are 3 SPA-3000s registered with Asterisk.
This approach relies on the SPA denying a call if it is already in use.
Looking through the logs, the SIP packets seem to be in order.
INVITE,
100-Trying, 504-Service Unavailable, ACK.
I'm at the end of my technical limit (ever increasing as I play in the
open-source world) - but my best guess is:
During the Register process, something is temporarily reset (such as a
variable indicating that the line is in use) such that when the second
call
comes in - it is actually connected to the existing conversation for a
brief
period before the SPA realizes that the line is actually already in use.
As part of a cleanup procedure - a hangup procedure is run:
disconnecting
the call.
The Equipment my trials were done on:
SPA3000 Hardware Version: 2.0.1(7376), Software Version: 3.1.10(GWd),
and also tried Software 3.1.7.
Nothing plugged into the FXS port.
Asterisk 1.2.4 running on
RE: [Asterisk-Users] Announcement System for a Charity
Yes, this is possible, but a management nightmare. -Original Message- From: bails [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Announcement System for a Charity Yes its possible, just create different contexts for each organisation. Bails Michiel van Baak wrote: On 13:40, Thu 20 Apr 06, Douglas Garstang wrote: Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. No, that is not possible with AMP/freepbx One of the reasons why I trashed it :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
sorry for my english, I did not explain myself correctly. I mean I downloaded the file Today, never meant to say that the file was uploaded Today. I know the file is recent enough because i looked for a change in mfcr2.c source that I know was put there recently. Regards On 4/22/06, Anton Krall [EMAIL PROTECTED] wrote: Are you sure its from today? The file has dates libmfcr2-0.0.3.tar.gz 30-Mar-2006 09:06 346K Also inside th tar the changelog has nothing inside and the news file has nothing too. How did you see it was from today? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Moises Silva |Sent: Saturday, April 22, 2006 9:21 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | |hum, the last time i downloaded something every file has |different dates. However, im looking at a new version that i |have downloaded |today: | |http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre9/ libmfcr2-0.0.3.tar.gz | |And checking the source it seems that tar is the most recent version. |I check the version looking in the C code for a fix i know |must be there, in mfcr2.c line 2780, after the generation tone |it must OR the signal with 0x80. | |Let me tell you that I have not tested that version. I have a |custom version that i fixed (because it gave me the same error |you have) and I sent the fix to Steve Underwood, but he told |me that my fix was not error proof, and that may fail (I have |1 month now in a production server with no problems tough), so |he made a similar fix, and told me that was more reliable. The |link I just gave you is for the TAR with Steve Underwood fix. | |I guess you already contacted me off-list to quote you for my |consultory. If you still have problems let me know and i may |be able to help you through SSH. | |Best Regards | |On 4/21/06, Anton Krall [EMAIL PROTECTED] wrote: | Moises, how can I find out which version Im running, on |Steves ftp all | say | 0.0.3 or the date also says the same date. | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf |Of Moises | |Silva | |Sent: Friday, April 21, 2006 9:43 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT. | | | |A couple of weeks ago, libmfcr2 has a small error in the tone | |signaling for the call setup, that was fixed 2 weeks ago or so, | |please, wich version of libmfcr2 are you using? if you dont |know try | |upgrading to the latest version. Im pretty much sure that you have | |the very same problem we had. | | | |Regards | | | |On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: | | Hello All, | | | | I'm facing problems with Unicall on this scenario : | | | | CentOS 4.3 - Running on x86_64 | | Asterisk 1.2.7.1 | | Zaptel 1.2.5 | | | | When running zttool , shows all Spans OK. | | | | But I can't receive and make calls. | | | | I tried to change many parameters and still doesn't work. | | | | Any clues ? | | | | * unicall.conf | | | | [channels] | | | | language=br | | | | context=incoming-pstn | | usecallerid=yes | | hidecallerid=no | | immediate=no | | callwaitingcallerid=yes | | threewaycalling=yes | | transfer=yes | | cancellforward=yes | | callreturn=yes | | echocancel=yes | | echocancelwhenbridged=yes | | | | rxgain=0.0 | | txgain=0.0 | | faxdetect=both | | loglevel=255 | | protocolclass=mfcr2 | | protocolvariant=br,20,4 | | protocolend=cpe | | group=1 | | callgroup=1 | | | | channel = 1-15 | | channel = 17-31 | | channel = 32-46 | | channel = 48-62 | | channel = 63-77 | | channel = 94-108 | | channel = 110-124 | | | | * zaptel.conf * | | | | loadzone=br | | defaultzone=br | | | | | | span=1,1,0,cas,hdb3 | | cas=1-15:1101 | | cas=17-31:1101 | | | | span=2,0,0,cas,hdb3 | | cas=32-46:1101 | | cas=48-62:1101 | | | | | | span=3,0,0,cas,hdb3 | | cas=63-77:1101 | | cas=79-93:1101 | | | | span=4,0,0,cas,hdb3 | | cas=94-108:1101 | | cas=110-124:1101 | | | | | | | | * lor error * | | | | -- Executing Dial(SIP/1000-1de2, | | Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 | | unicall_report: MFC/R2 | | UniCall/1 Call control(1) | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Make call | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: | | MFC/R2 | | UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 | | WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 | | UniCall/1 0001 - [1/ 1/Idle /Idle ] | | -- Called g1/40020022 | | Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 |handle_uc_event: | | Unicall/1 event Dialing | | Apr 20 19:13:57
Re: [Asterisk-Users] some EICON Diva 4BRI questions
2006/4/24, Armin Schindler [EMAIL PROTECTED]: When using a card with onboard DSPs (or even the software fax of AVM Fritzbinary-only driver) you can do faxing with the CAPI interface. That meansyou don't get the audio data stream, you get the fax-data instead which can be save in a file.In that case the application (Asterisk or anything else) don't need to dothe fax processing, this is done by the driver (in case of AVM Fritz) or onthe hardware in case of DSPs (like Eicon DIVA Server). chan-capi supports this and if the CAPI driver/device supports fax overCAPI, you don't need anything like spandsp. That's what I mean with faxingover CAPI.ArminThanks for all. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
My telco used cat5 as well for the demarc to CPE. It's also with noting that many channel banks, such as my Atlas, and zapata.conf itself also have parameters to allow you to tune the gains to compensate for cable signal loss. I've never had to touch them, and my CPE is about 300 feet from the PRI demarc (with an ordinary Cat5 cable connecting them) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outbound calls to sip urls
On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly: Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. I'm no expert, but it looks simple enough to me - just use the originate action to call with something like this: Action: Originate channel: SIP/[EMAIL PROTECTED] context: testcontext extension: extensiontosendtheprompt priority: 1 So that extension will just send the prompt and then hang up. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User Defined VoiceMail announcement?
Benoit Panizzon wrote: Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Option 0 when logging into VoicemailMain() to check your messages. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
Yes it is possible - check out the Asterisk manual or nice book from O'Reilly - Asterisk PBX (The Furute of telephony) Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hi...Please help me
For hardware check out this page: http://www.digium.com/en/products/hardware/ Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement System for a Charity
when did you trash it since they changed to freepbx they have added a new permissions based login and they have also split out users and devices. Very nice for setting up a company with different divisions. You can give the support extensions to the support manager to deal with.It is well worth the upgrade to freepbx On 4/21/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 13:40, Thu 20 Apr 06, Douglas Garstang wrote: Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. No, that is not possible with AMP/freepbxOne of the reasons why I trashed it :)--Michiel van Baakhttp://michiel.vanbaak.info [EMAIL PROTECTED]GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2DWhy is it drug addicts and computer afficionados are both called users? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- $15.95/Month DreamHost Hosting SALE60 GB Disk Storage, 1.6 TB TransferTransfer Increases 16 GB WeeklyUse discount code caralena for $40 off all these low prices ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help!!!!! DTMF detection is not working on Zap lines
Hi all, I am running 1.2.7.1 asterisk on FC3. Every thing works except dtmf detection on my Zap lines. I am using a TE411P with isdn NI2. Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asteriak not starting with Ground Start Lines
Eric ManxPower Wieling wrote: Davi-Ann wrote: When I set asterisk to to sequence the lines as Ground Start the system is not starting. It is giving the following error Invalid Argument 22 Do you have any ideas about this. Any help or assistance appreciated. I don't think Digium's analog cards support Ground Start.. Though Digium gives inconsistent answers, the TDM400 FXS module DOES provide a Ground Start trunk, which can be used as an interface into another PBX with that requirement I also have not yet tried this with the latest Zaptel drivers. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue reload
On 4/24/06, Johann [EMAIL PROTECTED] wrote: I've noticed that when app_queue.so is reloaded(or just a reload command is used) that all queue members that were paused are automatically unpaused. Is there a workaround for this? (Note, I use statically defined callback agents). That sounds like a bug. Please post a bug on bugs.digium.com. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Alexander Lopez wrote: I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. Good thing he doesn't work for a cable manufacturer as that's a total crock of crap that even an inexperienced person should be able to detect. (You can't twist two wires to make them weight three times as much, or cost three times as much.) He told me that with the number of twists in the Cat 5 cable it would cancel out any interference, but he also stated that the effective length was calculated using a cable with less twists and subsequently 'less dense' and that if using a Cat5e cable you must factor that in. so if you use cat5e cable your are fine but you can't go as far. Essentially true, but the impedance of a T1 cable is different from Cat5 cables, which is one of the primary factors in limiting distance. Has nothing to do with the twists. Shielded vs non-shielded has to do with the environment, and how much electrical noise there is near the T1 cable. Nothing more, nothing less. Regarding the Smart Jack it is mostly used as a location at the CPE where the Telco can loop and make sure that the problem is at your end. So your assumption is correct that you can plug anything you want into it, its one your side of the demark, so if it doesn't work it's YOUR problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User Defined VoiceMail announcement?
RTFM On 4/24/06, Benoit Panizzon [EMAIL PROTECTED] wrote: Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the Voicebox of John Smith, please leave a message after the tone). Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
On Mon, 24 Apr 2006, Rich Adamson wrote: Alexander Lopez wrote: I was once told by a lineman that the cables they use didn't have that many twists in them because it wasn't needed, and that the extra twists would effectively use more cable and thus cost and weigh more than triple what they do now. Good thing he doesn't work for a cable manufacturer as that's a total crock of crap that even an inexperienced person should be able to detect. (You can't twist two wires to make them weight three times as much, or cost three times as much.) A foot of cat5 has more than 12 on each of the individual wires inside. Not much but there is some difference. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P support on POTS around the world (Slovakia)
Hello everybody, does anybody use P100P FXO card on POTS lines in Slovakia, Bohemia (Czech rep.), Poland, Hungary...? I need to know if those cards work especially in Slovakia or if you can reccomend FXO cards for Slovak POTS lines. Thanks, Marcel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, April 22, 2006 5:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax and URA
Hi, I have an URA that says to my customers: "Dial 1 for support, dial 2 for sending a fax". This URA starts with g729, but when the call is transferred to the RxFax, it should be converted into g711, for the fax to work. Is there a way to solve this??? Thank you!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossovercable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to aTE410P ?
Good thing he doesn't work for a cable manufacturer as that's a total crock of crap that even an inexperienced person should be able to detect. (You can't twist two wires to make them weight three times as much, or cost three times as much.) He may have started out as an underground lineman, posibly inhaling too much swamp gas and CO from passing cars, I took his coments at face value. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fxotune Problem
Hi, Well, I have a big problem with Asterisk, my problem is that when I'm in a conversation, using zap channels, in a moment the line has a interferce that produce a sound in the conversation, this sound is a electratical sound I think, I was reading about that and I found that the utility fxotune can help me to change some settings about the audio and the supression of interferences. My problem is that fxotuno does not work, I execute this script like that: #./fxotune -i 4 The output of this command is: /dev/zap/1 absent: No such device or address /dev/zap/2 absent: No such device or address /dev/zap/3 absent: No such device or address /dev/zap/4 absent: No such device or address /dev/zap/5 absent: No such device or address /dev/zap/6 absent: No such device or address /dev/zap/7 absent: No such device or address /dev/zap/8 absent: No such device or address /dev/zap/9 absent: No such device or address /dev/zap/10 absent: No such device or address /dev/zap/11 absent: No such device or address /dev/zap/12 absent: No such device or address /dev/zap/13 absent: No such device or address /dev/zap/14 absent: No such device or address /dev/zap/15 absent: No such device or address /dev/zap/16 absent: No such device or address /dev/zap/17 absent: No such device or address /dev/zap/18 absent: No such device or address /dev/zap/19 absent: No such device or address /dev/zap/20 absent: No such device or address /dev/zap/21 absent: No such device or address /dev/zap/22 absent: No such device or address /dev/zap/23 absent: No such device or address /dev/zap/24 absent: No such device or address ... Reading the source code I found that this error is produced with system call O_RDWR. My system is running [EMAIL PROTECTED] 2.5 with Astersk 1.2.4 and Zaptel 1.2.5 compiled, with a Digium Wildcard TDM2400P with 24 FXO Modules. I don't understand why fxotune can found /dev/zap/ devices. I hope that someone can help me. Thanks for yor help, Roly Morales ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850?
Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Faster Sound Files
I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? Thanks Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Essentially true, but the impedance of a T1 cable is different from Cat5 cables, which is one of the primary factors in limiting distance. Has nothing to do with the twists. Shielded vs non-shielded has to do with the environment, and how much electrical noise there is near the T1 cable. Nothing more, nothing less. I always love these discussions on cat5 vs T1 cable. cat5 is NOT T1 cable and if any telco/vendor tried to install it in my location I'd have them pull it and put in the proper cabling. T1 cable is not just insulated cable, the cable pairs are separately insulated, not just for enviroment conditions but to prevent cross talk. The only safe way to try to use cat5 cable as a T1 cable would be two runs of cat5, one for Tx and one for Rx. It is necessary for the Tx and Rx signals to be in separate sheaths to prevent cross talk interferance. guess that's just me thou. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
How about using LVS? http://www.ultramonkey.org/3/topologies/lb-overview.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: lunes, 24 de abril de 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, April 22, 2006 5:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. No confidentiality or privilege is waived or lost by any wrong transmission. If you have received this message in error, please immediately destroy it and kindly notify the sender by reply email. You must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. Opinions, conclusions and other information in this message that do not relate to the official business of Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it. -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. I want to have the SIP HEADER like this: FROM: sip:CALLERID(number)@domain.tld thanks best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850?
Thanks for the hints and tips. While you are familiar with the 2850, I am using the PERC raid controller but guess this shouldn't make any real difference. I used the middle PCI slot for the TE210P, do you use any particular slot. I will disable HyperThreading and the box was already running an SMP kernel (there were no irq conflicts shown by lspci -v) in runlevel 3. Are you using the onboard e1000 ethernet controllers? The wiki is advising not to. Thanks for your input! Remco On Mon, 24 Apr 2006, Craig Guy wrote: Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
Well, for a start, there's a single director, which means a single point of failure. Really, I wonder why they even bother. -Original Message- From: Sergio García Murillo [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers How about using LVS? http://www.ultramonkey.org/3/topologies/lb-overview.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: lunes, 24 de abril de 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, April 22, 2006 5:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers Although there maybe a better way, this would work: 1. Add the IP's into your sip.conf and set qualify=yes. 2. Make your dialplan something like the following: exten = _X.,1,Dial,SIP/[EMAIL PROTECTED] exten = _X.,2,Hangup exten = _X.,102,Dial,SIP/[EMAIL PROTECTED] exten = _X.,103,Hangup exten = _X.,203,Dial,SIP/[EMAIL PROTECTED] exten = _X.,204,Hangup exten = _X.,304,Dial,SIP/[EMAIL PROTECTED] exten = _X.,305,Hangup This would make your failover work but certainly wouldn't help with the load balancing between the servers. If any cannot qualify or are congested, they will automatically failover to the next server. I believe most people use an SER proxy for this type of application. It seems to work well with the round robin type DNS. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hill Sent: Saturday, April 22, 2006 5:13 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Connecting to a cluster of SIP servers My Asterisk server is connecting to sip.plus.net, which resolves to multiple IP addresses: sip.plus.net. 300 IN A 84.92.0.75 sip.plus.net. 300 IN A 84.92.0.76 sip.plus.net. 300 IN A 84.92.5.189 sip.plus.net. 300 IN A 84.92.5.190 If one of these machines is down (i.e. it's not replying to the SIP packets or it's sending back ICMP Port Unreachable), Asterisk keeps trying the same server. Shouldn't Asterisk move on to the next server automatically in this case? It seems to only way to do this at the moment is to run the reload command, which causes it to do a DNS lookup and it may then pick one of the other servers. -- - Steve xmpp:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1447 (20060316) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. No confidentiality or privilege is waived or lost by any wrong transmission. If you have received this message in error, please immediately destroy it and kindly notify the sender by reply email. You must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. Opinions, conclusions and other information in this message that do not relate to the official business of Ydilo Advanced Voice Solutions, S.A. shall be understood as neither given nor endorsed by it. --
[Asterisk-Users] getting listed in Directory Assistance, the phone book
Has anyone had any luck getting listed in directory assistance when your number is ported from For example, I have an asterisk box for a client, that is also shared with another client in the same building. The CLEC provided PRI and numbers (including the ported #s from Verizon) as the main owner of the PRI (call them Tenant A) and the CLEC will not change the listing name for Tenant Bs numbers. How does this process work? Is there some other company I can contact to get this information updated or is it entirely dependant on each phone book provider? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
On Monday 24 April 2006 11:42, Ken Godee wrote: cat5 is NOT T1 cable and if any telco/vendor tried to install it in my location I'd have them pull it and put in the proper cabling. T1 cable is generally Cat3 is it not? That's certainly how the old T1s loops were run between the CO and the business... From the smartjacks I've seen shielded Cat3 but Cat3 nontheless. T1 cable is not just insulated cable, the cable pairs are separately insulated, not just for enviroment conditions but to prevent cross talk. Insulation (especially such thin insulation) does not prevent crosstalk. Distance, shielding and tighter twists do. The only safe way to try to use cat5 cable as a T1 cable would be two runs of cat5, one for Tx and one for Rx. It is necessary for the Tx and Rx signals to be in separate sheaths to prevent cross talk interferance. Unless you're going for some kind of distance record, standard Cat5 will work without any issue on any modern installation. As I said, I'm pretty sure (not 100%, but close) that the T1 specification is only Cat3, since it's standard BellCore wire and they don't run your T1 loops (which aren't T1 anymore, they're DS1 over HDSL or HDSL2) in special high end trunks. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
On Mon, 24 Apr 2006, Douglas Garstang wrote: You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. Huh? Has this happened to you in practice? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMP kernel on Pent 4?
Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from scratch. I installed FreePBX (CentOs) from scratch and asterisk was running, but had not yet been configured. It too crashed with a kernel panic. Ran memtest for 24 hours; no errors or issues uncovered. I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) Nothing in the logs to suggest a root cause and I'm now waiting on recurrence using the non-SMP kernel. R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
[EMAIL PROTECTED] wrote: Where can we find a roadmap of asterisk 1.4 release ? Harry... please use proper mailing list etiquette when posting to these lists. It is very tiresome to see you quote an entire long message, without changing the subject, and insert a one-line unrelated comment at the top of your reply. To answer your question: there is no roadmap for 1.4. We just began the 'scheduled release' cycle with this release, and we are still trying to feel our way into the process and learn how much work we can accomplish in a release cycle. Once 1.4 is done, I expect we will be putting together a roadmap for 1.6, although given that the project gets a great deal of its code from volunteer contributions, putting something on a roadmap is in no way any guarantee that it will be part of that release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
Rich Adamson wrote: Oh, and if shielded T1 cable is used, the shield at each end of the cable must be grounded. (Let's see how many can figure out how to do that via an rj45 plug. ;) You use shielded plugs and jacks, of course :-) That is why the TE405P/TE410P have shielded jacks (as of about a year ago, IIRC). The retail packaged cards even ship with four shielded cables included! Minor point: isn't it safer to only ground the shield on one end? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Andrew Kohlsmith wrote: Insulation (especially such thin insulation) does not prevent crosstalk. Distance, shielding and tighter twists do. Ever looked at the underground cable in the street outside your building? If it's more than 20 years old, it's probably paper-insulated gel-filled cable, with an _extremely_ thin amount of insulation between the conductors and _zero_ insulation between the pairs. T1s seem to work just fine on it, unless it's very old or they try to put more than 6-8 spans in a single 100-pair bundle :-( ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMP kernel on Pent 4?
Well, are you running an SMP box, or is it just hyperthreaded? I know there are issues with running an SMP kernel on a machine that's only HT. On Mon, 24 Apr 2006, Rich Adamson wrote: Had a Pent 4 server running fc3 crash (kernel panic) and am rebuilding from scratch. I installed FreePBX (CentOs) from scratch and asterisk was running, but had not yet been configured. It too crashed with a kernel panic. Ran memtest for 24 hours; no errors or issues uncovered. I then noticed that FreePBX installed using a SMP kernel (and grub indicated a non-SMP kernel was installed as well). Would running an SMP kernel on a Pent 4 potentially cause a kernel panic? (Or, do I need to dig somewhere else?) Nothing in the logs to suggest a root cause and I'm now waiting on recurrence using the non-SMP kernel. R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)
Thomas Winter wrote: Hi, I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name) for the initial INVITE to an SIP proxy. If I use SET(CALLERID(name)=) the display-name is asterisk. Just a guess, try: SET(CALLERID(name)= ) Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
Unless you're going for some kind of distance record, standard Cat5 will work without any issue on any modern installation. As I said, I'm pretty sure (not 100%, but close) that the T1 specification is only Cat3, since it's standard BellCore wire and they don't run your T1 loops (which aren't T1 anymore, they're DS1 over HDSL or HDSL2) in special high end trunks. And if you don't believe the 'high-end' part brush up against a 66-block while your well grounded, you will be singing in the high-end!!! At that voltage I think the differential created by the twists would cancel anything including small animals out along the line. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Monday, April 24, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Connecting to a cluster of SIP servers On Mon, 24 Apr 2006, Douglas Garstang wrote: You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. Huh? Has this happened to you in practice? It sure has. Polycom phone queries DNS for domain.com and gets round robin IP of 192.168.10.1. It sends a REGISTER request to that IP. Asterisk at 192.168.10.1 sends back a 407 Proxy Auth required. The polycom phone then queries DNS again and gets 192.168.10.2 this time and sends the REGISTER with auth info included this time to Asterisk at 192.168.10.2. The Asterisk at 192.168.10.2 box goes 'huh. What the hell is this for?' becuase it never received the original REGISTER, and drops it on the floor. The phone never gets an OK to its register request. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting to a cluster of SIP servers
On Monday 24 April 2006 11:12, Douglas Garstang wrote: You can't use round robin DNS. Round robin DNS will cause every SIP packet to potentially go through a different static path, which will break things. Um... The media gateways do not do a DNS lookup for every packet they send out... At least no sane one... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Andrew Kohlsmith wrote: On Monday 24 April 2006 11:42, Ken Godee wrote: cat5 is NOT T1 cable and if any telco/vendor tried to install it in my location I'd have them pull it and put in the proper cabling. T1 cable is generally Cat3 is it not? That's certainly how the old T1s loops were run between the CO and the business... From the smartjacks I've seen shielded Cat3 but Cat3 nontheless. No, cat3 isn't the same. But, for short distances and no induced noise (as stated previously) anything will do, even jumper wire. The T1/E1 interface spec's are typically 75 ohm balanced (BNC, E1), 100 ohm balanced, etc. I've never bothered to check to see if cat5 cables use the appropriate mating twisted pairs or not. Since the pinouts are different for cat5 vs T1 cables, I'd have to guess a single strand is used from two different twisted pair groups. That wouldn't be cool, but in short runs it probably doesn't have much of an impact. R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
On Monday 24 April 2006 12:13, Kevin P. Fleming wrote: Minor point: isn't it safer to only ground the shield on one end? Yes, you *never* shield both ends. That can cause ground loops and add to the long list of what the..? head-scratching problems that telephony has. As to WHICH end to ground... well that's a subject that holy wars have been started over... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Faster Sound Files
On Mon, Apr 24, 2006 at 09:41:40AM -0600, Douglas Garstang spake thusly: I'd like to increase the speed of the Asterisk sound files. Miss Alison talks a bit slow. I can use sox to increase the speed, but then the pitch changes and she starts to sound like a chipmunk. Any audio experts out there know how I can increase the speed a little bit, and change the pitch accordingly so that she sounds ... normal? I think the 'stretch' command in sox is what you need. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, April 24, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ? Rich Adamson wrote: Oh, and if shielded T1 cable is used, the shield at each end of the cable must be grounded. (Let's see how many can figure out how to do that via an rj45 plug. ;) You use shielded plugs and jacks, of course :-) That is why the TE405P/TE410P have shielded jacks (as of about a year ago, IIRC). The retail packaged cards even ship with four shielded cables included! Minor point: isn't it safer to only ground the shield on one end? ___ If you want to fry your cards attach both sides to ground, I don't know about the engineering specs but 'floating' one side has always eliminated ground-loops, and the possibility of lightning damaging the cards. As a rule I always ground the side furthest from my equipment as I want the lightning to go that-a-way!!! Electricity is lazy by nature and always tries to find the shortest (least resistive) path to ground. I make sure that my CPE uphill for the spike.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
Ever looked at the underground cable in the street outside your building? If it's more than 20 years old, it's probably paper-insulated gel-filled cable, with an _extremely_ thin amount of insulation between the conductors and _zero_ insulation between the pairs. T1s seem to work just fine on it, unless it's very old or they try to put more than 6-8 spans in a single 100-pair bundle :-( 6-8 spans? That's the number that I have been trying to get, and why the limit. Is it X-talk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about Asterisk realtime
Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed that there is no ip address stored for my softphone in Mysql, how does the Asterisk know which computer my softphone is running? I checked the config files, no softphone registrations in sip.conf. Did I miss anything to configure the system? Thanks for your help. Tielin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID/variable setting.
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten = _XXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1})
exten = _XXX,3,NoOp(${CALLERIDNUM})
exten = _XXX,4,Dial(${OUTBOUNDTRUNK}/${EXTEN})
(I wanted to test against my own extension, 1625; if that worked, I
wanted to strip off the 1, and then prepend the 603-123-4 to my
remaining three digits.)
Which is all well and good -- until I actually try to use it. Then, I get:
-- Executing GotoIf(SIP/1625-f89a, 0?4:2) in new stack
-- Goto (internal,7654321,2)
-- Executing Set(SIP/1625-f89a, CALLERIDNUM=6031234625) in new stack
-- Executing NoOp(SIP/1625-f89a, 1625) in new stack
-- Executing Dial(SIP/1625-f89a, Zap/g1/7654321) in new stack
Why does my NoOp line show my 1625 extension, when CALLERIDNUM is -- as
far as I can tell -- being set to 6031234625? (I looked against the Set
command page on the Wiki, and I think I'm doing it right.)
Asterisk 1.2.3, if that matters.
Thanks,
-Ken
___
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Re: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Ok, Im not a developper but what do you think of both a wish list . Harry To answer your question: there is no roadmap for 1.4. We just began the 'scheduled release' cycle with this release, and we are still trying to feel our way into the process and learn how much work we can accomplish in a release cycle. Once 1.4 is done, I expect we will be putting together a roadmap for 1.6, although given that the project gets a great deal of its code from volunteer contributions, putting something on a roadmap is in no way any guarantee that it will be part of that release. ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting to a cluster of SIP servers
Huh? Has this happened to you in practice? It sure has. Polycom phone queries DNS for domain.com and gets round robin IP of 192.168.10.1. It sends a REGISTER request to that IP. Asterisk at 192.168.10.1 sends back a 407 Proxy Auth required. The polycom phone then queries DNS again and gets 192.168.10.2 this time and sends the REGISTER with auth info included this time to Asterisk at 192.168.10.2. The Asterisk at 192.168.10.2 box goes 'huh. What the hell is this for?' becuase it never received the original REGISTER, and drops it on the floor. The phone never gets an OK to its register request. Doug. Makes sense. We only use the round robin records for the outbound proxy, so the only time they use the other servers is when they make outgoing calls. As for the registrations, the phones register with a primary server and a secondary server (or in the case of the polycoms, if server one is down, it re-registers with server two immediately... it works, we've tested it), so any server in the round robin group knows about the phones when they make outbound calls. However, I think in the sense of actually MAKING phone calls, once it gets an ip address for a round-robined host, it continues talking to that host... i.e. Once the phone's in a conversation, all related packets for the stream go through the same server, not to any other server in the group. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Kevin P. Fleming wrote: Andrew Kohlsmith wrote: Insulation (especially such thin insulation) does not prevent crosstalk. Distance, shielding and tighter twists do. Ever looked at the underground cable in the street outside your building? If it's more than 20 years old, it's probably paper-insulated gel-filled cable, with an _extremely_ thin amount of insulation between the conductors and _zero_ insulation between the pairs. T1s seem to work just fine on it, unless it's very old or they try to put more than 6-8 spans in a single 100-pair bundle :-( Those paper insulated cables are still the best. The ones laid in the 1930s are still like new. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
On Monday 24 April 2006 12:39, Alexander Lopez wrote: And if you don't believe the 'high-end' part brush up against a 66-block while your well grounded, you will be singing in the high-end!!! At that voltage I think the differential created by the twists would cancel anything including small animals out along the line. It's only 130VDC, at moderate (but not high by any means) current. This is 26AWG, after all. -A. (only... I work in industrial power, in fact there's 575VAC at my desk where I'm typing...) The medium voltage (4160VAC) stuff is out back. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
On Monday 24 April 2006 12:42, Rich Adamson wrote: The T1/E1 interface spec's are typically 75 ohm balanced (BNC, E1), 100 ohm balanced, etc. Ahh yes, this is true. Is that a typical spec for even POTS lines? I've never bothered to check to see if cat5 cables use the appropriate mating twisted pairs or not. Since the pinouts are different for cat5 vs T1 cables, I'd have to guess a single strand is used from two different twisted pair groups. That wouldn't be cool, but in short runs it probably doesn't have much of an impact. Well Cat5 doesn't specify pinouts at all, just frequency response and crosstalk between pairs, IIRC. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover
Rich Adamson wrote: I've never bothered to check to see if cat5 cables use the appropriate mating twisted pairs or not. Since the pinouts are different for cat5 vs T1 cables, I'd have to guess a single strand is used from two different twisted pair groups. That wouldn't be cool, but in short runs it probably doesn't have much of an impact. They are fine, actually, other than color-coding difference. Since Ethernet-on-twisted-pair standards were derived from existing telco standards for cabling, the pairing in the 8-position connector is compatible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
I've never bothered to check to see if cat5 cables use the appropriate mating twisted pairs or not. Since the pinouts are different for cat5 vs T1 cables, I'd have to guess a single strand is used from two different twisted pair groups. That wouldn't be cool, but in short runs it probably doesn't have much of an impact. IIRC, standard Ethernet uses pairs 12 and 36. The color scheme on 568B is 12 = white/orange pair, 36 = white/green pair Most Ethernet cables then have the white/blue pair on 45, and white/brown on 78. An RJ45 carrying a T1 is: 1 - RxA 2 - RxB 4 - TxA 5 - TxB Assuming that you'd want RxA and TxA in the same twisted pair (ditto for RxB and TxB) then a cable would look something like this at each end: 14 = white/orange pair 25 = white/blue pair I don't know if there's an industry standard for T1 cabling to have a certain color pair for A and another for the B pairs. Electrically, though, the color is insignificant - as long as the correct pairs are twisted together then all is well. Does anyone have a real T1 cable that they can share with us the pin configuration? I am curious to know what color pairs are used. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
Alexander Lopez wrote: 6-8 spans? That's the number that I have been trying to get, and why the limit. Is it X-talk? I think so. I've had clients before who had to have spans brought in via different routes even though the pairs in the underground cable were in otherwise acceptable condition. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to Linphone sound playback delay, and then choppy
Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound keypressing the pound key...) echoed rather well, but sound quality is a little poor. I have Sipomatic tested over the same cross-over network connection... perfect. I have made calls to my mobile via SipGate... takes a while to start, but then perfect. I thought it might be because I was using OSS emulation, but I recompiled to use ALSA pure (without 'Jack', that had issues) and got the exact same results. Does anyone have a working config they could post, or any idea what may be the issue? I am running Asterisk on a 1600Mhz laptop, so I doubt it's short of cycles - and I don't think I did anything special as regards config. I used this as a template for the linphone config http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+linphone I've tried the apt-get asterisk (1.2 I think) on Debian, and also this PXE version (booting my laptop from the board) http://www.automated.it/asterisk/pxeindex.html And this as a starting point for my Asterisk config files. http://www.automated.it/guidetoasterisk.htm#_Toc49248767 Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails)
When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and hangs up before the voicemail starts to record, an empty message will still be recorded and sent to the user. I've tried setting minmessage=3 in voicemail.conf, but that only seems to work when calling from SIP phone to SIP phone - for calls through the PSTN, it makes no difference (since it seems the Zap channel isn't disconnecting immediately). I'm also curious about the maxsilence setting in voicemail.conf, it says: Maxsilence defines how long Asterisk will wait for a contiguous period of silence before terminating an incoming call to voice mail. The default value is 0, which means the silence detector is disabled and the wait time is infinite. Maxsilence takes a value of zero or a positive integer value which is the number of seconds of silence to wait before disconnecting. does that mean that once a call goes to voicemail, asterisk will wait maxsilence seconds before disconnecting the call, even if the user is still connected? I guess this is for handling the case where the dialing in user is still connected, but hasn't actually said anything, ie, they may have had to rush away from the phone for some emergency without hanging it up first, so rather than have asterisk record everything, it disconnects after maxsilence seconds have elapsed with no sound.. If this is the behaviour, then it looks like maxsilence has nothing to do with preventing a dropped call from being recorded in the first place, since it seems that the maxsilence behaviour is only applicable after voicemail has already picked up. So if anyone has any other suggestions, please let me know, since I'm getting a lot of complaints from users about empty voicemails, and would like to fix this as soon as possible. Thanks in advance, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
