[asterisk-users] Trying to configure xorcom on Suse 11
I've never seen e4e4:1164 before.
What does this output?:
lsusb|sed -e 's/:/ /g'| grep e4e4| awk '{print astribank_tool -n -D
/proc/bus/usb/$2/$4}'| bash
reset the astribank:
#(if you use freepbx)
amportal stop
#(if you start asterisk that way)
/etc/init.d/asterisk stop
/etc/init.d/dahdi stop
/usr/share/dahdi/xpp_fxloader reset
#give it time
sleep 30
/usr/share/dhadi/xpp_fxloader load
#(you should see e4e4:1162)
lsusb
#(you should see the hardware here)
dahdi_hardware -v
#presuming you have /etc/dahdi/system.conf right this will work
/etc/init.d/dahdi start
#if you use freepbx
amportal start
#or
/etc/init.d/asterisk start
--
JD Austin
Twin Geckos Technology Services LLC
j...@twingeckos.com
Voice: 480.288.8195x201
Fax: 480.406.6753
http://www.twingeckos.com
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[asterisk-users] Trying to configure xorcom on Suse 11
Xorcom hardware uses three layers; you must resolve issues in the following order: 1. USB 2. Dahdi 3. Asterisk I suspect you're having trouble with the usb layer. Run lsusb It will display a line like this if the firmware isn't loaded: Bus 001 Device 004: ID e4e4:1161 If it is e4e4:1162 then the firmware is loaded. You can manually load the firmware like this: /usr/share/dahdi/xpp_fxloader load or /usr/share/dahdi/xpp_fxloader usb -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Custom Extension Timeout
[EMAIL PROTECTED] wrote: Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using [EMAIL PROTECTED] 2.8 and a TDM400P card. Please, any help is greatly appreciated! Craig I'm running Asterisk 1.2.12.1 and Freepbx 2.1.3 and have this problem also. Also on a TDM400P card. I've tried setting up a queue, ring group, followme, none of the timeouts are obeyed. JD ___ --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] zap channel media volume
I've been struggling with this issue for over a year. I wish there were some kind of automatic gain control built in to set the rx/tx gain on the fly based on the volume of the two channels. Probably not realistic though. Is there other hardware other than digium's that better deals with this issue? Rich Adamson wrote: The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this 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 ___ --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] zap channel media volume
I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? regards, Wolfgang ___ --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] fax
That's a vague question Khaled :) First you must have hardware/software to support fax. Faxes on my TDM400P work great with asterisk; they didn't work so great over voip or with my X100P though. Next you need software.. spandsp works ok to me with my fax to email setup. Read all about it here: http://www.voip-info.org/wiki-Asterisk+fax JD Khaled Chehab wrote: How can I support fax at trixbox M. Khaled Chehab Monitoring Operationg Engineer Xplorium Tel: +961 1 868686 Fax: +961 1 808810 e-mail: [EMAIL PROTECTED] * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Asterisk Quit Registering with Broadvoice
Joe McConnaughey wrote: Hello - I've been using Broadvoice with Asterisk for a couple of months with no issues. Today, it has stopped registering. The Sip Debug from CLI is below. It tries to register five times and then gives up. Any suggestions? As you might suspect, I have not been able to get Broadvoice on the phone and usually get cut off after being on hold about 5 minutes. I would be VERY surprised if it was your setup that was the issue. I recently dumped them as a provider after several months of 'iffy' service. Check their BOYD setup page for asterisk in case they've changed something there, other than that crossing your fingers will do as much good as trying to contact them. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
Erik Espinoza wrote: All I'm saying is avoid sixTel. They have great sound quality, good prices, good ping times, and a pretty control panel. However if you ever need support anything done that doesn't happen automatically on their web site (number doesn't ring, custom toll free did, refund, support in general) forget about it. It's been three months since i ordered my custom toll free did, and it's not active yet! Calling their support number just tells you to go to the web site. Adding tickets, even emergency tickets, seem to go to /dev/null. Also if anyone knows of a provider that does LNP, would be able to move my numbers away from sixTel if they don't respond after a certain date and has decent prices please let me know. Erik That mirrors exactly the experience I had with them. I even got them on IM a few times and they said 'give me an hour and I'll have it fixed'.. never happened. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So you all think VoIP sypply is warm and fuzzy
Michael D Schelin wrote: Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I don't here back from you shortly and receive the configuration program that should have shipped, I will return it back to you for a full refund and express my views to the Voip community. As of now I've herd of nothing but good things about your customer support. I've called and left messages to your support team. I waited 7 days for this unit and have no way to configure it. Email me the CD. Michael D. Schelin Owner Shelltel Are you sure you didn't buy a refurbished model? I hear they sell a lot of refurbished equiptment, I've purchased some of it myself. Everything I've purchased from them worked without issue. None however came with an installation CD. A few things had to be reset to clear settings though. Anything I needed was freely available. Since you know how to contact Medatrix, perhaps you can download the software or get a CD from them. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk perl radiusclient
It's complaining that you don't have the perl module installed or it is not in your path. Kamran Ahmad wrote: hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten = _X.,1,agi,agi-rad-auth.pl|Routing=SIPAuthorizeBy=SIP vi /etc/asterisk/modules.conf load = res_agi.so ---errors *CLI Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.5/i386-linux-thread-multi /usr/lib/perl5/5.8.5 /usr/lib/perl5/site_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.4/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.3/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.2/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.1/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.5 /usr/lib/perl5/site_perl/5.8.4 /usr/lib/perl5/site_perl/5.8.3 /usr/lib/perl5/site_perl/5.8.2 /usr/lib/perl5/site_perl/5.8.1 /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.4/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.3/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.2/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.1/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.5 /usr/lib/perl5/vendor_perl/5.8.4 /usr/lib/perl5/vendor_perl/5.8.3 /usr/lib/perl5/vendor_perl/5.8.2 /usr/lib/perl5/vendor_perl/5.8.1 /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl .) at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. BEGIN failed--compilation aborted at /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10. Jul 6 19:38:54 WARNING[30695]: app_dial.c:516 dial_exec: Dial argument takes format (technology1/number1technology2/number2...|optional timeout) Sell on Yahoo! Auctions – no fees. Bid on great items. http://auctions.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Emergency Asterisk Guru Help needed EMERGENCY
My /var/spool/asterisk has the following directories drwxr-xr-x2 asterisk asterisk 4096 Jul 5 10:55 fax drwxr-x---2 asterisk asterisk 4096 Jul 4 18:53 monitor drwx--2 asterisk asterisk 4096 May 11 17:10 outgoing drwxrw2 asterisk asterisk 4096 Mar 31 05:26 qcall drwxr-xr-x2 asterisk asterisk 4096 Mar 31 05:23 tmp lrwxrwxrwx1 asterisk asterisk 37 Jul 4 20:16 vm - /var/spool/asterisk/voicemail/default drwxr-xr-x3 asterisk asterisk 4096 Jul 4 14:34 voicemail drwxr-xr-x2 asterisk asterisk 4096 May 11 17:10 wakeups I'm running [EMAIL PROTECTED] 1.3 though. JD Jeffrey Starin wrote: 911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory structure under /var/spool/asterisk? Thanks, B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does Vonage support fax machines?
You can use the NVDetect stuff from Newman Telecom works with some success (I had it working). http://www.voip-info.org/tiki-index.php?page=NVFaxDetect I think echo cancellation can play a factor too. It isn't perfect though.. I had some page smearing, especially on the first page but it worked ok. They probably don't compensate for packet loss. Their service isn't perfect either: http://www.complaints.com/directory/2004/december/28/52.htm JD Deon wrote: My boss is insisting we support fax, and I keep telling him that Fax over IP is very unreliable and not recommended and his immediate come-back is Vonage does it. and it's very hard to figure out how. I don't think Vonage does T.38, the Linksys/Sipura units they're using doesn't support T.38 to my knowledge. That means they have to be using G.711Ulaw to send faxes. But how do they compensate for packet loss/jitter/etc. In our test lab, the best we could get was 90% success at sending faxes. It seemes to screw up the longer the transmission, ie page 1 was usually ok, but page 2 and 3 and 4 was at serious risk. So if I bought a Vonage adapter, can I send a 30 page fax? My best guess is they have high quality voice T1's, like from an ILEC, usually more expensive, and when they sell a Fax Line I noticed it's more expensive. Maybe they route all their fax calls specifically out these high quality T1's that they own, so that they can do some type of quality control. My test lab was a private network, a Cisco 3640 connected to a local voice PRI T1, and converting to SIP. Asterisk would push the calls to the Cisco 3640 and the Linksys PAP2 would register with Asterisk. All local. I then tried several test faxes throughout the PSTN. Would it be better to plug the voice T1 straight into Asterisk using one of Digium's cards? Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Epia C3 Linux
Tried knoppix? Wiley Siler wrote: OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Did it require any special work or did you just download the ISO for FC3 and install? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 Linux Anyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? Thanks Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failover question
John Cianfarani wrote: What if asterisk was to start have more options for failover from an application perspective? Eg. Some form of heartbeat between the two servers. Within the heartbeat it could pass registration information and call information between servers. (Not sure if this is somehow possible already) So if you were to use that with something like HA/clustering the backup server would always know what calls / registrations were active. Thanks John A hot thing in databases right now are clusters. Has anyone setup a linux cluster and installed asterisk on it? -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?
New features include: CentOS 3.5 Asterisk 1.0.8 New Zaptel Driver from CVS Built-in DHCP server David Shaw wrote: Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SixTel?
I was just checking out the dids for all of my fail over providers and noticed that neither DID that I have with SixTel work. Both pause for a long long time The local number gives a recording: 'The number you have dialed is not in service or is assigned in a different area code. Please check your number and dial again'. The 800 number just rings busy. Anyone else having this issue or am I a lone data point? JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Asterisk and Mambo - help wanted
We do mambo projects all the time, contact me off lists and we can get you rolling. JD Kristian Kielhofner wrote: Hello everyone, So, this isn't exactly what it seems. I am not looking to integrate Asterisk and Mambo. I am the maintainer/creator of AstLinux, and I have recently decided that I should really have a better web site for it. I would like to use Mambo so that I can do updates easily, from anywhere, without having to waste time learning PHP/HTML/etc. Mambo CMS seems the best and most powerful way to do this. It's not that easy, however, to go from the default Mambo site to a site suitable for an open source project such as AstLinux. I just need some help to get a layout, theme, etc. going. Updates and maintenance I can handle (probably). So, what I am looking for is someone who is familiar with Mambo (and preferably Asterisk, too) and would be willing to help me jump start astlinux.org/.com. Because AstLinux is an open source project, I will be unable to directly compensate anyone (monetarily) for their work at this time. However, any people that help out are more than welcome to plug their own projects, companies, names, etc. on the site (within reason). Interested? Comments? Questions? Suggestions? Drop me a line. Thanks! -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Cron and Reload
Heres what I use; I have asterisk restart if the net is down and once a
day just before 7am:
crontab -e:
55 6 * * * /usr/sbin/asterisk -r -x restart gracefully /dev/null 21
5,10,15,20,25,30,35,40,45,50,55 * * * * /root/scripts/check_net
/root/check_net
#!/usr/bin/perl
$net=`/bin/ping -c 02 google.com 21 | /bin/grep -c 'unknown host'`;
if ($net==1) {
print `/bin/date`;
print `/usr/sbin/asterisk -r -x restart gracefully`;
}
Federico Alves wrote:
This will sound weird but the command 'asterisk -r -x reload' fails to work
when issued by Cron. But it works when I issue it from a bash session. What
is not configured correctly? I need to refresh the configuration every a
short amount of time.
rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005
Date: Wed, 15 Jun 2005 18:42:00 -0400
From: [EMAIL PROTECTED] (Cron Daemon)
To: [EMAIL PROTECTED]
Subject: Cron [EMAIL PROTECTED] asterisk -r -x reload
X-Cron-Env: SHELL=/bin/sh
X-Cron-Env: HOME=/root
X-Cron-Env: PATH=/usr/bin:/bin
X-Cron-Env: LOGNAME=root
/bin/sh: line 1: asterisk: command not found
Any ideas?
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--
JD Austin
Twin Geckos Technology Services LLC
email: [EMAIL PROTECTED]
http://www.twingeckos.com
phone/fax: 480.288.8195
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Re: [Asterisk-Users] rxfax not answering
rxfax doesnt work with voip, you need something like NVFaxDetect from Newman Telecom to detect the incoming fax. Essentially you sent him an email and he'll send you the code. Once you compile them into asterisk you can add it. http://www.voip-info.org/tiki-index.php?page=NVFaxDetect JD Antonio Gallo wrote: Hello i would like to receive incoming faxes thru' asterisk as tiff files thru' the rxfax application. I setup extensions 101 like this exten= 101,1,rxfax(/tmp/fax.tif) then from CLI i run: dial 101 and rxfax send me his scream about the fax ^^ instead when i send a real fax from a faxmachine to that extension my 101+rxfax is executed but it just does nothing the call is originated by a FAX on PSTN and received via VoIP by asterisk using a/u law codec i think that is my VoIP provider that has some fax problem. Is this the problem or there maybe other solutions? Thank you, Antonio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help! Zap echo on bridged calls
I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . Inbound calls through the X100P that do not bridge to voip are fine. Outbound calls that do not bridge with the X100P are fine. PSTN -*-VOIP calls have so much echo on the called party side (sidetone) that it is almost impossible to have a conversation. I've played with rxgain txgain, echocancelwhenbridged,etc nothing seems to work. Yes Im completely restarting asterisk and running ztcfg. Anyone figured this out? Heres my zapata.conf : ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=1.0 txgain=3.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help! Zap echo on bridged calls
I should note that echocancelwhenbridged=yes was tried. JD Austin wrote: I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . Inbound calls through the X100P that do not bridge to voip are fine. Outbound calls that do not bridge with the X100P are fine. PSTN -*-VOIP calls have so much echo on the called party side (sidetone) that it is almost impossible to have a conversation. I've played with rxgain txgain, echocancelwhenbridged,etc nothing seems to work. Yes Im completely restarting asterisk and running ztcfg. Anyone figured this out? Heres my zapata.conf : ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=1.0 txgain=3.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice - Customer feedback
Luki wrote: Can any broadvoice customers give me their opinions on the service recently? In short: it sucks, at least for me. I not one who's quickly complaining about them, but the audio quality for US calls has been not acceptable since the fiasco a few weeks ago. The audio is very choppy on the PSTN side, regardless of proxy or number dialed (incoming and outgoing calls). International calls sound clear as they did before, no complains here. So call completion works fine, the phone rings but it doesn't mean you can actually hear something :-(. Yes, I'm parting from them step by step, with all accounts. If I could only find a reliable DID provider that offers numbers in my area... oh well. --Luki ___ That sums up my experience with them. They started out pretty good, now inbound calls dont work, outbound calls sound like crap. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Karl
I don't believe this type of rhetoric belongs on this list. Please take it somewhere more appropriate such as moveon.org, johnkerry.com , http://dean2004.blogspot.com/, or even http://www.rushlimbaugh.com. Now if you have a question about your Sipura3000 or dialplan/etc post away. JD Libel Lawyer wrote: This is the guy that has a ton of email addresses. Almost as many as he has phone numbers. google kvj He doesn't like our president either: SNIP -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crashing; Not getting Core dumps
Matthew Boehm wrote: This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGR -Matthew Turn on debugging in asterisk. It will write more to the log files. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN-voip/sip echo
JD wrote: I'm still relatively a novice with asterisk and am having issues with echo. The calling party that calls a PSTN number doesnt hear the echo, but the answered side via sip or forwarded to another PSTN number over voip hears excessive echo that makes it difficult to communicate. I've been playing with the zapata.conf settings for echocancel, echotraining, rxgain, txgain, etc and am basically stabbing in the dark (grin) I've read the wiki about it, but it doesn't go into very much detail. Anyone know which parameters fix this issue? Is there an easier way than tweaking settings in zapata.conf, monitoring with ztmonitor, and restarting asterisk over and over? JD I should add that it's only on calls that are bridged. Calls inbound to PSTN don't hear echo, sip/iax in/calls out don't hear echo but when they're bridged (inbound PSTN outbound VOIP) the called party hears echo badly. What can I do about it? JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LOOKING TO HIRE
Jean-Michel Hiver wrote: Preston Garrison wrote: I think what you want is a Senior Scripter not a Senior Programmer :) Perl, PHP, Python? I doubt any good programmer is going to want to use those scripting languages.. Excuse me sir, but you seem to know nothing about Perl or Python. Please refrain from talking bullshit about things you don't know anything about from now on. Best Regards, Jean-Michel. I concur with Jean; you can do some quite amazing things in perl and python in a month that would take you much longer in 'C'. It's too bad that programming language snobbery still exists. I used to think that Perl was too slow for production code.. but hardware nowdays more than makes up for any overhead Perl adds, especially if you design your application correctly. If you're running your app on a 486 with 64M of ram.. write it in assembly code. If you have a 3GHZ machine with a 2G of ram.. you have a lot of choices :) JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.288.8195 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup other ringing phone
http://www.asterisk.org/index.php?menu=support#handbook_project http://www.digium.com/handbook-draft.pdf Mark Brown wrote: Hi everyone, Is there a simple way of answering a different ringing extension from a sip phone using AAH? I have absolutely zero technical know-how when it comes to modifying conf files etc. Still working on figuring it all out. ;) That brings me to my second question... where the hell does one find an extensive manual of sorts that explains all conf files and what the strings all mean etc? Cheers All Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipSupply.com
I've ordered several things from them; all arrived as expected. Last time I ordered from voipsupply but the order was fulfilled by B2 TECHNOLOGIES LLC (same company I think). JD Manjit Riat wrote: I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if its a wrong post. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX jitter
Steven Langley wrote: Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is intermittent break up in the audio when they are receiving. I have had a look at iax2 show channels and for the first 2 users (those with no audio problems), the Jitter is low (0006ms), and the Lag is relatively high (00070ms). For the 3rd user (the one with audio breaking up), the Jitter is relatively high (0627ms) and the Lag is relatively low compared to the others (00012ms). All of the users are on the same LAN, so cant quite understand the differences. Any explanations / ideas would be much welcome. Thanks Steven I'm having the same issue; I'd love to know how to actually diagnose the issue. I assume there is somewhere on the voip-info.org wiki about this? JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live Voip
They're really slow to set things up. I signed up 2 nights ago.. still waiting. I'm going to give the the benefit of the doubt.. but am signing up for other voip services in the mean time. JD My question to them: It's been over 24 hours. I figured out how to login, my accountcode appears to be X. The DID I selected is not attached to the account as far as I can tell. I would think this sort of thing would be more automated. JD Their response: We are not automated. The reason is security. Even credit card charges are all verified. Its all to protect the system from abuse. If we did not verify things the amount of people who would attempt fraud would be un-real. Our techs are just working alot of orders and do everyone in the order they arrived. Sales LiveVoip LLC Sean Kennedy wrote: Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk at home with Broadvoice?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Disgruntled Asterisk Luser Sent: Sunday, May 08, 2005 9:06 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Asterisk at home with Broadvoice? Don't worry about these subtle details. Broadvoice has been off the air for almost a solid week, with no real explanation as to what the problem is. On the voxilla.com board, there have been a lot of inferences, but no real solid information as to what the problem actually is. But I really wonder about the survivability of a telephony service business which is unusable for days on end, and with no explanation of what the problem is. My connection is back up.. but I just noticed an issue.. DTMF is iffy. Numbers I call that previously worked fine (press *, enter access code) can't recognize the numbers I enter. I think it's time to switch, question is.. to who!? JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] livevoip
Anyone use livevoip? opinions? -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Issues
No, you're not the only one.. Lets see.. current count is down 3 times in 10 days. ***SIGH*** So, the question is.. what voip providers are reliable right now? JD Sean Milheim (iDREUS Corporation) wrote: Anyone else having problems with Broadvoice? While trying to dial I receive a recording We're sorry this call can not be completed at this time. Also Can't call in on BV number. Unfortunately this is becoming a common occurrence with them. -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Issues
My outbound calling just started working.. inbound still broken. JD Kerry Garrison wrote: Same here using LAX proxy. All circuits are busy dialing out Fast busy calling in -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Milheim (iDREUS Corporation) Sent: Thursday, May 05, 2005 7:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice Issues Anyone else having problems with Broadvoice? While trying to dial I receive a recording We're sorry this call can not be completed at this time. Also Can't call in on BV number. Unfortunately this is becoming a common occurrence with them. -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Issues
Think we'd like to see all of the itsp's succeed, and let their value-add services (including support) drive the market. However as a past customer, I don't see BV stepping up to the plate anytime soon. So, it really doesn't make any difference what we all want them to do, and its doesn't appear their going to support * with anything better then what we've all seen for the last six to twelve months. When one truly boils down what is received for BV service for their price, BV is no where near a special/good deal. Several other itsp's are lower price and have higher quality support, even though most still don't offer the level of support that we'd all like to see. Too many are still using the isp support model right now. That too will change over time. So who are those itsp's? I went with Broadvoice because I'd heard good things about them, at this point they're one more outage from losing me as a customer. I was fine for the first week with them, but I've had micro outages and outages that last for hours for the last few weeks. Im ready to jump to something more reliable. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question PSTN-VOIP forwarding and # of inbound calls
If I get a standard business line from qwest, plug it into my FXO card and get the call forward busy service, will that allow me to handle more than 2 inbound calls? If I can do that, then I can use that for inbound calling and switch to any voip provider I want for outbound calling. One caveat is callerid.. I'd have to block it since the # is different unless theres a way to set the name on the voip provider. All my outbound calls say 'Phoenix'. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Fwd: Call forwarding]
Works for me, though I believe asterisk just bridges the call. JD Michael D Schelin wrote: As far as I know Asterisk does not support normal PSTN type call forwarding. I.E. the user would type *72 etc. This is called call forking. My Mulitech gateway does but at a huge price. Also T38 is supported. I have several carriers that I use that have Asterisk. All of the Asterisk boxs won't accept call fowarding. I send the calls to my carriers with Cisco gateways and the calls reroute correctly. Now I have a proxie that controls everything. You may be able to do call fowarding with 2 boxes. But a call in and reroute back out may not work. Damian Funnell wrote: Any takers? Sometimes the most basic questions yield the least replies, huh? Cheers, Damian. Original Message Subject: Call forwarding Date: Wed, 04 May 2005 08:40:41 +1200 From: Damian Funnell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com References: [EMAIL PROTECTED] [EMAIL PROTECTED] Hi team, Basic question I know, but I can't seem to find any obvious information about this: Does anyone know if * natively supports call forwarding from a given extension (i.e. call forwarding without having to write a macro)? My user wants to be able to dial a code plus a phone number to start diverting all calls to the given extension to that number. Call forwarding would then be disabled by dialling a code number again. I expected that * would support this type of feature natively, but can't find anything in the wiki. If responding please let me know if we need to enable anything in features.conf as well. Thanks in advance, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Andre Normandin wrote: Hello, About 4PM EDT I noticed that my broadvoice service cannot register.. Anyone else having problems with their broadvoice service? FYI: I connect to the 147.135.20.128 (nyc) proxy... Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD My connection is back up. Maybe they DO read this list :) -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] automated availabilty testing
Has anyone figured out how to know if your voip connection goes down without actually dialing in and out manually? During all of the broadvoice outages I've had as far as asterisk was concerned, I was registered. It was only when I tried dialing in or out that it became obvious that something was up. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
Max Clark wrote: Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max Nope, I get the same thing. I can dial out though through my asterisk machine, but not in from pstn. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
I tried calling Broadvoice support.. on hold for 1/2 hour then it hung up on me with a reorder (fast busy), so I tried again. Just got through to a rep- they said it's a 'carrier issue' that their 'partner carrier' was having issues and that it would be up soon. Makes me wonder if I should be signing up with their 'partner carrier' instead. JD Jerry Geis wrote: I am having the same broadvoice issue at the moment. jerry Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
I guess 2 hours is 'soon' to them. I'm still down. Is there a reliable voip provider out there that works with Asterisk? I can't have downtime like this.. it just makes my company look bad. JD JD Austin wrote: I tried calling Broadvoice support.. on hold for 1/2 hour then it hung up on me with a reorder (fast busy), so I tried again. Just got through to a rep- they said it's a 'carrier issue' that their 'partner carrier' was having issues and that it would be up soon. Makes me wonder if I should be signing up with their 'partner carrier' instead. JD Jerry Geis wrote: I am having the same broadvoice issue at the moment. jerry Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
Sean Kennedy wrote: Actually, with all the threads I've seen on the mailing list, I'm weary of anything having to do with broadvoice. Personally. Maybe it's just that they have such a large user base on linux. Who knows. Voicepulse gets my business tho. :) Sean Voice pulse doesn't offer dids in my state (AZ) :( -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
Looking at asterisk from the command line I notice that there is stuff after my number; -hex#; I've never noticed this before. Is it normal? Why would this suddenly stop working? I haven't touched anything on the server for a few days. JD Connected to Asterisk 1.0.7 currently running on asterisk1 (pid = 2269) Verbosity is at least 4 -- Executing GotoIf(SIP/4804221250-dd40, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/4804221250-dd40, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/4804221250-dd40, ) in new stack -- Executing NVFaxDetect(SIP/4804221250-dd40, ) in new stack -- Executing GotoIf(SIP/4804221250-bd48, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/4804221250-bd48, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(SIP/4804221250-bd48, ) in new stack -- Executing NVFaxDetect(SIP/4804221250-bd48, ) in new stack -- Executing Hangup(SIP/4804221250-dd40, ) in new stack == Spawn extension (from-pstn-reghours, h, 1) exited non-zero on 'SIP/4804221250-dd40' -- Executing Hangup(SIP/4804221250-bd48, ) in new stack == Spawn extension (from-pstn-reghours, h, 1) exited non-zero on 'SIP/4804221250-bd48' asterisk1*CLI quit Chuck Smith wrote: I am having trouble with Broadvoice as well. My server sees the call but I get no audio then the line drops but the call stays up at the asterisk server. Goes to voicemail but on the far end the phone is already on the hook. Glad I wasn't the only one. Thought I was going crazy over here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Monday, April 25, 2005 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice Down? Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
On Mon, 2005-04-25 at 12:48, JD Austin wrote: I guess 2 hours is 'soon' to them. I'm still down. Is there a reliable voip provider out there that works with Asterisk? I can't have downtime like this.. it just makes my company look bad. JD I just came up.. 4 hours down time. There are a few changes on broadvoices' web site: http://www.broadvoice.com/support_install_asterisk.html I made the changes, but they didn't immedately work so I don't think that was it. What does 'pedantic=no' mean in sip.conf? JD JD Austin wrote: I tried calling Broadvoice support.. on hold for 1/2 hour then it hung up on me with a reorder (fast busy), so I tried again. Just got through to a rep- they said it's a 'carrier issue' that their 'partner carrier' was having issues and that it would be up soon. Makes me wonder if I should be signing up with their 'partner carrier' instead. JD Jerry Geis wrote: I am having the same broadvoice issue at the moment. jerry Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
Kerry Garrison wrote: While we have only been using Broadvoice for a few months now, we have actually had better service through them than with our PSTN provider. You could just as easily have had a voice T1 go down which typically takes a few hours to replace (and may be the actual problem). The issue is not with Broadvoice as a service as much as it seems to be a peering problem. Yes these issues make you and your company look bad in so much as it shows that you do not have a properly designed redundant system. Do you back up your server every night even though you only lose a file once a year? So why wouldn't you have a failover on your phone system? Its very easy to blame Broadvoice for your phone service being out (and I am not defending them) but that is ONE connection. Any business that is highly reliant on their phones for business should have a backup system just for this reason. Possibly even multiple backups depending on the critical nature of the business. I have seen businesses completely lose telephone service on standard PSTN lines for a day or more at a time. So sh** happens, be prepared. Dont put all your eggs in one basket. This is a perfect example of why. Kerry Garrison http://techdatapros.com I do have failover; I am using another service that is not voip based. Fortunately only a few clients know about the new BV number. This makes me reluctant to switch over completely. How exactly could you failover using such a provider? Outbound calling failover is easy; just have another provider and set them up as a trunk. Inbound which relies on them and the DID provided to work is a bit more complicated. Sure you could do the same, but other than 'call this other number if this number is down' how do you failover? The only way I can think of is get a regular pots line in and have it forward on busy to BV and have both answer in asterisk. Are DID's portable? JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip problems
Kerry Garrison wrote: The quick and dirty method is to have the main lines come in over PSTN lines, this can then be call-forwarded to a VOIP provider. This makes it simple to change VOIP providers on-the-fly as well as turn off forwarding and let the PSTN line come into the PBX. Setting up failover on outbound calls through multiple providers is relativly simple but keeping the main business number pointing to whatever service you are using is the tricky part. -Kerry Doesn't that limit you to a single inbound call at a time? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gerard Marcel Sent: Monday, April 25, 2005 1:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voip problems How do you guys deal with voip problems? do you have multiple backups such as land lines, and different voip providers? Regards, GM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Email
Its an AMP issue. Log into AMP, Click Maintenance Click Config Edit. Click vm_email.inc Scroll over and you'll see the IP address you want to change. JD Chris wrote: Im running [EMAIL PROTECTED] 0.8 so I dont know if this pertains solely to the @home version or if you guys can help me but When I get a voice mail it sends an email with a link to the voicemail system. Problem is the IP address is wrong and I need to change it. Christopher Dittrich, There is a new voicemail in mailbox 202: From: "SMITH KENT D" xx Length: 0:48 seconds Date: Wednesday, April 13, 2005 at 12:54:33 PM Dial *98 to access your voicemail by phone. Visit http://192.168.1.101/cgi-bin/vmail.cgi?action=""> to check your voicemail with a web browser. The IP is suppose to be 192.168.3.200, but I dont know where to look. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I retrieve voice mail in Asterisk
Depends on your dial plan. for me it's *98 then enter mail box number then password. JD Chuck Bunn wrote: Hi, I guess I am dense or something but I cannot figure out how to retrieve voicemail using a SIP SJPhone or and Analog phone with Astyerisk. I googled (voicemail +retreive :lists.digium.com) and did not get much. Everything works. I can ring each extension and if it doesn't answer it goes to voice mail, but I can't figure out how to retrieve it. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice
Im curious about that too.. if so how many concurrent calls will they allow? JD Matt wrote: Hi, I'm currently routing my asterisk server out over broadvoice.. it seems I can do multiple outgoing and incoming calls does anyone know if broadvoice actually allows this or not? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Up @Home 0.8 Guide
Much appreciated Kerry! JD Kerry Garrison wrote: Because of all of the changes to AMP, we have written up a completely new How-To Guide for [EMAIL PROTECTED] v0.8. Our first example uses BroadVoice for the trunk. http://www.geekgazette.com -Kerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.6 - Release Date: 3/30/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting Up @Home 0.8 Guide
Would imbedding it in an alt tag work. I've always been curious about what actually works to help blind users navigate/etc. JD hank smith wrote: is there a way you can write those screen shots in to text format on the user guide? I am a blind computer user and am unable to see the examples that are shown on the site. thanks hank - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:12 PM Subject: [Asterisk-Users] Setting Up @Home 0.8 Guide Because of all of the changes to AMP, we have written up a completely new How-To Guide for [EMAIL PROTECTED] v0.8. Our first example uses BroadVoice for the trunk. http://www.geekgazette.com -Kerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.344.2640 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.6 - Release Date: 3/30/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Third party Firefly issue very weird??
First guess.. firewall. Jon Walsh wrote: When I connect to the third party softphone (firefly) I get connected at my house and at my office where I have the asterisk..but when I went to my friends house to set him up his firefly showed a gray circle like it was not connecting at all? Has Anyone seen this happen what is causing this no to connect, does anyone know ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie pointers
Kerry your site was a godsend for me, thank you! JD Kerry Garrison wrote: warning - shameless plug We have been writing some How-To guides and will be doing different product reviews as well. So far, we have had a very good response. Check out our site you will find some things to help get your started. http://www.geekgazette.com -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Blaise Sent: Thursday, March 24, 2005 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie pointers Hello all I have come to Asterisk with no previous telco experience. As I will be playing with Asterisk really soon, I would like to have some pointers as to some tutorials in telco that could help me get into all this. I am quite a beginner, don't forget :) Thanks a lot! Best, fred ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home version 0.6 forwarding to pstn numbers?
I was hoping I could set up extensions and do *72XX to get them to ring at outside numbers. When I do it though it just sits there silently for quite a while and then hangs up. Is there a special way to do this? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
xlite doesn't seem to have this problem. Sys Admin wrote: well i even pressed ctrl+alt+del went into the process monitor and gave the firefly process high priority. Still it looses half a second of sound each time i maximize or minimize a app like putty, whats the word for this . sucks . why doesnt skype have this problem ? t On Thu, 24 Mar 2005 10:39:54 -0800 (PST), Robert Hajime Lanning [EMAIL PROTECTED] wrote: Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who="Sys Admin" On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * - SMS w/out PSTN (dialplan confusion)
From what I understand the voice mail is not actually sent to the user
via email.. a link to it is.
They have to log in and listen to the message. If you delete the voice
mail it probably won't work.
I'm a newbie but I think that is the problem.
JD
Mark Charlton wrote:
Hi
I tried to setup this system, and tried to follow their, bayhamsystems,
instructions for installing the app version of the util, but there was no
makefile.patch in the apps source folder so that failed, (I don't really
understand the make business on linux anyway), so I tried to use the perl
AGI version. My query about priority would apply to all sms providers I
assume.
I installed and configured the perl script as instructed, then came to set
up the dialplan. My dial plan is below, based on the one from bayhamsystems
at http://www.bayhamsystems.com/asterisk.html. I couldn't figure out how
theirs worked as if you hung up after leaving the voicemail their call to
the AGI script never executed, so I modified it as below.
However I have problems that I don't understand.
1) on h,1 the hasnewvoicemail command should add 101 to the priority, what
does it do if there is no mail? What does priority h,2 do? They had it going
to the equivilent of priority +98. Is that just so it hangs up if it gets an
invalid priority goto? The wiki entry on hasnewvoicemail is very quiet on
what happens if there is no newmail.
2) if a caller left a voicemail, then the hasnewvoicemail returned true and
went to h2,102 however if the second caller rang, and hung up BEFORE leaving
a message, since the first callers message was still waiting, another call
to the AGI was made, to notify me of a message I was already aware of,
(incuring extra cost). I changed the voicemail.conf to
attach=yes|delete=yes, (since I was emailing the vm anyway), and now the
hasnewvoicemail always returns false. Please suggest any alternative
approaches.
3) the AGI returns with exit code 0 as in completed successfully, however I
don't see any texts, nor does my credit count decrease at bayhamsystem
account mangement page. However a test message sent from their web site
arrived at my phone within 2 minutes of being sent.
;;;
;; snip from dial plan
;;;
exten = 0,1,goto(sales|s|1)
exten = _[123],1,SetVar(dialed_extn=${EXTEN})
exten = _[123],2,Dial(SIP/200SIP/202|2|m)
exten = _[123],3,Playback,hold1
exten = _[123],4,Dial(SIP/200SIP/202|2|m)
exten = _[123],5,playback,all-busy
exten = _[123],6,Voicemail(30${EXTEN})
exten = _[123],7,playback,thank-call
exten = _[123],8,Hangup
exten = h,1,HasNewVoiceMail(30${dialed_extn})
exten = h,2,goto(h,100)
exten = h,102,DeadAGI(fastsms|447803xx|Caller ${CALLERID} left a new
voice mail at ${DATETIME} on Sales extn ${dialed_extn)|asterisk)
exten = h,103,Hangup
-Original Message-
From: William Suffill
http://www.bayhamsystems.com/ has a app for sending SMS with asterisk.
Don't know how their prices stack up for the UK though.
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[Asterisk-Users] Settings to improve voice quality?
Im using Broadvoice and just got it working last night. Once noticable annoyance is that the audio quality is pretty poor. There are pops and volume fading. Are there settings that will improve this? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setup to dial out only on voip (Broadvoice) not PSTN?
I've been trying to get a new asterisk box setup with Broadvoice for over a week now. I have it connecting and registering with them according to 'sip show registry', I can't dial out through it, but it does dial out through my regular phone line. I'd like to set it only to dial 911 through that line and have all other calls go over voip. I've checked out a bunch of viop-info pages, anyone already setup with Broadvoice that can help me out? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setup to dial out only on voip (Broadvoice) notPSTN?
Kerry Garrison wrote: Do you have the broadvoice trunk set as the Default Trunk? -Kerry Looks like I have more reading to do :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JD Austin Sent: Tuesday, March 22, 2005 8:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Setup to dial out only on voip (Broadvoice) notPSTN? I've been trying to get a new asterisk box setup with Broadvoice for over a week now. I have it connecting and registering with them according to 'sip show registry', I can't dial out through it, but it does dial out through my regular phone line. I'd like to set it only to dial 911 through that line and have all other calls go over voip. I've checked out a bunch of viop-info pages, anyone already setup with Broadvoice that can help me out? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please for newb on Asterisk to Vonage
Im a newbie to this list (joined today). Other than Broadvoice, what voip providers work well with Asterisk? I'd like a service that will allow trunking so that I can have more than one outbound/inbound call if possible. JD Kerry Garrison wrote: You arent going to make this happen as you describe. Vonage is not a good service to use with Asterisk. To quote from the Wiki: Vonage service is locked to the ATA they send you. It is not possible to connect Asterisk (or any other SIP UA) directly to your main Vonage service. http://www.voip-info.org/tiki-index.php?page=Asterisk%20and%20Vonage If you want to use Asterisk, you will need a different provider. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J W Sent: Tuesday, March 22, 2005 11:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help please for newb on Asterisk to Vonage I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
