RE: [Asterisk-Users] Broadvoice / * re-register issues
Kevin's entry in sip.conf does not have caller id properly defined NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Digium T100P T1 Card
Peter, I also made it a point to voice my appreciation and recognize the fact that Stephen is major contributor here. I also acknowledged his generous explanations. I have also since replied to his reply and thanked him again as well. A consultant so I can get a T1 PRI on my wall and use it with my Asterisk box? LMAO. That is the dumbest thing I have ever heard. I need a consultant so I can get a T1 with PRI? Please. I am just trying to better understand how the Digium PRI card works and how it interconnects to the ISP. I checked the Wiki and I checked Digium. Neither one said install PRI card and no other router is needed. Or rather, what I did find was the reference that said that your * box will act as a router with the PRI card. Then it clicked and I got it. Having never had a PRI T1, I did not know it would be unlike my current T1 which has an AdTran to break out my voice from the data. So asking how to connect the Digium card seemed natural for this discussion. Again, thank you for you contribution to the discussion and for offering your view. If my previous response was offensive to anyone, especially Stephen, I apologize. If it is not clear, I view the gurus here as generous contributors. I just generally don't like to feel criticized or spoken down to when I am just asking a simple honest question. Isn't that the point of this all? I mean, it is not like I am asking how to insert a PCI card or something. Maybe that is the price to pay but really I think it just is not usually needed. It would be so much easier to just not respond if what I have sent is so horrible. Sometimes I think that this response is just because so many questions are asked and people get tired of poor formatting and such. Regardless, my OWA was to blame for part of that. Speed typing was the other. Besides, asking how my T100P interconnects is hardly fishing for general telecom knowledge. The questions were specific to the hardware from Digium. That makes it pretty relevant I think. I am googling up other stuff mentioned by Stephen right now. Thanks all, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Wednesday, January 05, 2005 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Digium T100P T1 Card On Wed, 5 Jan 2005, Wiley Siler wrote: So, I need to learn more about voice T1s? Reeally? That would be why I am posting to the user group in the first place. To learn more. The wiki says nothing about how PRI works because it is expected that someone will know. Well, I didn't. Had to ask. After cruising ebay for 30 minutes looking at routers and reading the tech spec on the T100P, I figured out the very same thing regarding the fact that no router was needed. [snip] However, was there that much need for the criticism and arrogance in your reply? Wouldn't it just be esier not to reply at all than start off with a complaint about my HTML formatting, go to a critique of how I formatted my 4 sentence email (paragraph for 4 sentences?), and finish up by pointing out that I don't know much about voice T1s? Normally I can be quite critical of the sometimes brusque replies on this list but the reply Steven sent was filled with information. He started out by saying that he found your email hard to read and the reasons why. He then stated that you have a lot to learn about T1/isdn pri which is probably true. This is a complex subject and if you are not familiar with it it may be a good idea to hire a consultant who is. This list is really not meant as a general educational tool for digital telecom. There are such resources elsewhere on the net. Once you have done your homework and is more knowledgeable on the topics of telecommunications you are in a better position to ask questions regarding Asterisk. At that point you will probably receive a lot more help from the members of this list. Peter ___ 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] Last callers script?
See http://www.wheely-bin.co.uk/asterisk/ check this link - I've implemented it and it works, at least in the test environment. John On Wed, 5 Jan 2005 16:00:56 +, Mike Dent [EMAIL PROTECTED] wrote: Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike ___ 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] Digium T100P T1 Card
On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote: snip To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too much to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. Large Snip So to summarize: Currently you have a T1 from an ISP, this ISP is currently delivering 6 analog FXS phone ports and delivering fractional T1 internet access over the Ethernet ports on the Adtran 616. To help clear up an issue that may have confused others, all the lines you have are delivered in digital form the Adtran converts the 6 phone channels to analog. In theory (from reading mailing list not from personal exp.) the Adtran could be replaced by a Linux box with a T100P and Asterisk (probably without any config changes on the ISP end). Later; Tim ___ 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] Do Not Disturb
As well as allowing *xx to be dialed in your device dialplan, do you also have those codes set up in extensions.conf to do TheRightThing(tm)? (ie set a database flag that then gets checked by your call an extension macro to see if DND is activated or not?) Paul ___ 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 / * re-register issues
I think you might have to add the line below to [sip.broadvoice.com]: insecure=very I know that it's required for other services, and probably with broadvoice as well. /Anders Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register = [EMAIL PROTECTED]:X:[EMAIL PROTECTED] ice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=## context=default dtmfmode=inband canreinvite=no disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw ___ 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] TE410P card in an HP-Compaq DL380 G4 server
On Wed, 5 Jan 2005, Eric Bishop wrote: I will certainly try that. Please also let me know your progress.. Didn't help for me. I also tried removing one processor with no benefit. So I've now given up. Steve ___ 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] Music from Freeplay music included in * ??
Hi On the www.asterisk.org main page it says Music provided by Freeplay Music with a link - Where is the music in the *config? I cant find any supplied music - is there any? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Compiling Problem
I have this error compiling ASTCC: [EMAIL PROTECTED] astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate DBI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 46. BEGIN failed--compilation aborted at ./astcc.agi line 46. make: *** [install] Error 2 [EMAIL PROTECTED] astcc]# I have also installed asterisk-perl from http://asterisk.gnuinter.net/ but i get same error ... any idea?? Rafael -- rrgv PS: [EMAIL PROTECTED] asterisk-perl-0.08]# perl Makefile.PL Writing Makefile for asterisk-perl [EMAIL PROTECTED] asterisk-perl-0.08]# make all cp lib/Asterisk/Manager.pm blib/lib/Asterisk/Manager.pm cp lib/Asterisk/Voicemail.pm blib/lib/Asterisk/Voicemail.pm cp lib/Asterisk/Outgoing.pm blib/lib/Asterisk/Outgoing.pm cp lib/Asterisk/QCall.pm blib/lib/Asterisk/QCall.pm cp lib/Asterisk.pm blib/lib/Asterisk.pm cp lib/Asterisk/AGI.pm blib/lib/Asterisk/AGI.pm Manifying blib/man3/Asterisk::Voicemail.3pm Manifying blib/man3/Asterisk::Manager.3pm Manifying blib/man3/Asterisk::Outgoing.3pm Manifying blib/man3/Asterisk::AGI.3pm [EMAIL PROTECTED] asterisk-perl-0.08]# make install Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk.pm Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk/Manager.pm Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk/Voicemail.pm Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk/Outgoing.pm Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk/QCall.pm Installing /usr/lib/perl5/site_perl/5.8.0/Asterisk/AGI.pm Installing /usr/share/man/man3/Asterisk::Voicemail.3pm Installing /usr/share/man/man3/Asterisk::Manager.3pm Installing /usr/share/man/man3/Asterisk::Outgoing.3pm Installing /usr/share/man/man3/Asterisk::AGI.3pm Writing /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi/auto/asterisk-perl/.packlist Appending installation info to /usr/lib/perl5/5.8.0/i386-linux-thread-multi/perllocal.pod ___ 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] Music from Freeplay music included in * ??
/var/lib/asterisk/mohmp3/ __ Dana Olson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Middleton Sent: Wednesday, January 05, 2005 2:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music from Freeplay music included in * ?? Hi On the www.asterisk.org main page it says Music provided by Freeplay Music with a link - Where is the music in the *config? I cant find any supplied music - is there any? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this message in error, please contact the sender and delete the material from any system. ___ 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] Digium T100P T1 Card
Tim, Thanks for the reply! Your expanation is correct. The AdTran delivers the FXS on the wall and is being converted from digital. I hope you are correct about the swapout and I will chase this up with ISP again. Originally, they told me that changing my service required making changes upstream and reprovisioning. Now I am beginning to wonder. So the equipment chain would look like this... Plug from NIU - Asterisk PRI Card - Out Onboard NIC to phones and data separately Is this accomplished via IPTables or does * do this? Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy Costello Sent: Wednesday, January 05, 2005 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium T100P T1 Card On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote: snip To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too much to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. Large Snip So to summarize: Currently you have a T1 from an ISP, this ISP is currently delivering 6 analog FXS phone ports and delivering fractional T1 internet access over the Ethernet ports on the Adtran 616. To help clear up an issue that may have confused others, all the lines you have are delivered in digital form the Adtran converts the 6 phone channels to analog. In theory (from reading mailing list not from personal exp.) the Adtran could be replaced by a Linux box with a T100P and Asterisk (probably without any config changes on the ISP end). Later; Tim ___ 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] Asterisk CPU priorities (nice?)
Should a watchdog be an internal part of the Asterisk core? The problem is generic. I.e. any real time process may swamp a machine, and therefor it is not Asterisk specific. arve5 This is a problem that can be solved in asterisk, though, with a watchdog, and/or something more elegant. I've implemented this in iaxclient (it gets used automatically if you use iaxclient as root on linux), and it should be done for asterisk as well. See http://bugs.digium.com/bug_view_page.php?bug_id=0003203 If this is implemented in asterisk, and asterisk swamps your machine for more than N seconds (where N is 4, for example), the real-timedness of asterisk goes away.. Arve5 Steve Kann wrote: Gilad Ben-Yossef wrote: Justin Carlson wrote: what is wrong with running asterisk with the -pg flags at startup? Which is exactly what I suggested: Since VoIP is a real time activity, simple nice really isn't enough. What you should do is mark the Asterisk proccess as a real time task for the Linux kernel to schedule accordingly. You can do this with Asterisk by passing the -p option to the Asterisk command line. And the warning still holds: A warning is due here: real time priority scheduled tasks are not something to be toyed with. You need to be root to be able to turn on this feature (meaning you have to be running Asterisk as root). A bug in Asterisk, a problem with mpg123 or a red alert on a FXO card can very well leave your system completly non responsive - so use with care. Having said that, I've been running an Asterisk server on a machine which is also used as SOHO firewall and file server for year now and it works great. This is a problem that can be solved in asterisk, though, with a watchdog, and/or something more elegant. I've implemented this in iaxclient (it gets used automatically if you use iaxclient as root on linux), and it should be done for asterisk as well. See http://bugs.digium.com/bug_view_page.php?bug_id=0003203 If this is implemented in asterisk, and asterisk swamps your machine for more than N seconds (where N is 4, for example), the real-timedness of asterisk goes away.. -SteveK ___ 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] Music from Freeplay music included in * ??
Hi, Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk from the digium web server - Whats the CVS command for a 'head' install ? Thanks On Wed, 5 Jan 2005 14:43:23 -0500, Steven Frazier [EMAIL PROTECTED] wrote: Hi John, Yes when you do the cvs head install, look in /var/lib/asterisk/moh -rw-r--r--1 root root 1939812 Jan 5 14:07 fpm-calm-river.mp3 -rw-r--r--1 root root 2582496 Jan 5 14:07 fpm-sunshine.mp3 -rw-r--r--1 root root 2217563 Jan 5 14:07 fpm-world-mix.mp3 [EMAIL PROTECTED] mohmp3]# Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Middleton Sent: Wednesday, January 05, 2005 2:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music from Freeplay music included in * ?? Hi On the www.asterisk.org main page it says Music provided by Freeplay Music with a link - Where is the music in the *config? I cant find any supplied music - is there any? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] ChanSpy - Should I repatch it ?
Julian, I'm also following this issue, so I guess you're not alone in the universe, even more I'm not sure why nobody's following this issue usefull as it seems. Anyway we'll probably start working on it soon if this happens I'll let you know. What I'm not sure is why this didn't make it to the CVS... bye, Matt - Original Message - From: Asterisk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, January 04, 2005 1:48 PM Subject: [Asterisk-Users] ChanSpy - Should I repatch it ? With the deafening silence from my previous questions, I feel seriously alone in the desire to have ChanSpy available. I want to be able to perform a ZapBarge on an Agents conversation, and ChanSpy was the answer to my prayers. Bug #2379 (http://bugs.digium.com/bug_view_page.php?bug_id=0002379) was closed bkw918 10-27-04 17:06 Closed pending new changes in cvs-head. These changes do not seem to be in CVS-HEAD as of today. Do I need to make the old patch work against current CVS-HEAD, or is it going to be available sometime ? I hope that I am not coming across as being awkward, I am more than happy to put in the work to make this patch work with CVS HEAD. I just don't want to do the work if someone else already is! Brian - I know that you are busy, please accept my apologies for any grief this may cause. Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew - Original Message - From: Muhammad Rizwan Khan [EMAIL PROTECTED] To: Asterisk-Dev@lists.digium.com Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL Hello I am trying to configure asterisk with MySQL for user authentication. According to dynamic friends To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS. This enables database definition of both IAX2 and SIP friends. Make sure you have the MySQL development kit (libraries) installed before compilation. http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers But in this make file, i did not find any disabled entry that need to be enable related to dynamic friends. Makefile is attached with email. plz. help me what should i need to do, to enable user authentication from MySQL. Thanks ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ 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] Forwarding Voicemail Crashes Asterisk
Hello everyone, As far as I can tell, if we try to forward a voicemail (by going into voicemail and saying that we want to forward it to another extension) it crashes asterisk. voicemail.conf does not seem to be where I should be looking. Any ideas? I did a 'cvs checkout -r v1-0_stable asterisk' when checking out from CVS. Should I be on a newer version? TIA, -Ron ___ 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] chan_oh323 gatekeeper
Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi ___ 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] Music from Freeplay music included in * ??
When you do a checkout, you will get 3 mp3 files that all begin with fpm- These are the 3 freeplay music files. -Matthew - Original Message - From: John Middleton [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Wednesday, January 05, 2005 1:47 PM Subject: Re: [Asterisk-Users] Music from Freeplay music included in * ?? Hi, Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk from the digium web server - Whats the CVS command for a 'head' install ? Thanks On Wed, 5 Jan 2005 14:43:23 -0500, Steven Frazier [EMAIL PROTECTED] wrote: Hi John, Yes when you do the cvs head install, look in /var/lib/asterisk/moh -rw-r--r--1 root root 1939812 Jan 5 14:07 fpm-calm-river.mp3 -rw-r--r--1 root root 2582496 Jan 5 14:07 fpm-sunshine.mp3 -rw-r--r--1 root root 2217563 Jan 5 14:07 fpm-world-mix.mp3 [EMAIL PROTECTED] mohmp3]# Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Middleton Sent: Wednesday, January 05, 2005 2:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music from Freeplay music included in * ?? Hi On the www.asterisk.org main page it says Music provided by Freeplay Music with a link - Where is the music in the *config? I cant find any supplied music - is there any? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Digium T100P T1 Card
On Wed, 5 Jan 2005, Wiley Siler wrote: Your expanation is correct. The AdTran delivers the FXS on the wall and is being converted from digital. I hope you are correct about the swapout and I will chase this up with ISP again. Originally, they told me that changing my service required making changes upstream and reprovisioning. Now I am beginning to wonder. So the equipment chain would look like this... Plug from NIU - Asterisk PRI Card - Out Onboard NIC to phones and data separately Look at the model number on the bottom of the adtran, if it says TDM it's channelized data, if it says ATM it's (surprise) ATM and Asterisk can't deal with it yet. -- 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 ___ 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] Allowing pooling or rollover for inbound calls on VoicePulse
My goal is to have only 1 primary phone number that can seamlessly pool multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to forward or roll over the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000 remains open for connections. Theoretically I would like to maintain 11 simultaneous connections all coming in from just the 212 555 1000 account. This situation applies if I wanted to voice conference more than 4 people on a single phone number (with multiple accounts on VoicePulse) Is this possible with Asterisk? ___ 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] Digium T100P T1 Card
Tim, Just confirmed with ISP that the NIU connects to the AdTran over HDLC. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Wednesday, January 05, 2005 12:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Digium T100P T1 Card Tim, Thanks for the reply! Your expanation is correct. The AdTran delivers the FXS on the wall and is being converted from digital. I hope you are correct about the swapout and I will chase this up with ISP again. Originally, they told me that changing my service required making changes upstream and reprovisioning. Now I am beginning to wonder. So the equipment chain would look like this... Plug from NIU - Asterisk PRI Card - Out Onboard NIC to phones and data separately Is this accomplished via IPTables or does * do this? Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy Costello Sent: Wednesday, January 05, 2005 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium T100P T1 Card On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote: snip To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too much to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. Large Snip So to summarize: Currently you have a T1 from an ISP, this ISP is currently delivering 6 analog FXS phone ports and delivering fractional T1 internet access over the Ethernet ports on the Adtran 616. To help clear up an issue that may have confused others, all the lines you have are delivered in digital form the Adtran converts the 6 phone channels to analog. In theory (from reading mailing list not from personal exp.) the Adtran could be replaced by a Linux box with a T100P and Asterisk (probably without any config changes on the ISP end). Later; Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: lcdproc and asterisk
Matt Riddell wrote: Corvin wrote: Hi! I would like to use lcdproc and asterisk. Any hints or links? Maybe someone has experience in such matter. I am working on such solution. I've heard of SAPBX. Thanks for any help. Hi, I was working with someone on this until my BB Forum fell over. Drop me a line if you need any help/have any questions regarding this. Hi, This is my first da on this project. I haven't it stared. Now I am preparing LCD and some documentation. So *any* help hint, program highly appreciate :). Corvin ___ 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] Digium T100P T1 Card
On Wed, 2005-01-05 at 12:45 -0700, Wiley Siler wrote: Tim, Thanks for the reply! Your expanation is correct. The AdTran delivers the FXS on the wall and is being converted from digital. I hope you are correct about the swapout and I will chase this up with ISP again. Originally, they told me that changing my service required making changes upstream and reprovisioning. Now I am beginning to wonder. So the equipment chain would look like this... Plug from NIU - Asterisk PRI Card - Out Onboard NIC to phones and data separately Is this accomplished via IPTables or does * do this? Asterisk would handle the voice portions, the data part would need to be handled first by the HDLC driver and then it will appear as an interface similar to a ethernet port. Then you would use IPTables or whatever your version of the kernel is supporting to act as a firewall as routing itself will be mostly natural once you let the kernel know to forward packets from one interface to another. While it is possible that there may not be any need to reprovision with the ISP, it is possible it would need to be too. Check to see if the data portion is being sent as HDLC. If so you should be able to discuss with them just replacing the Adtran box with your asterisk box. Then you just need to make provisions for your fax machines to get access to a phone line either via a SIP to FXS device or maybe one of the TDM400 cards. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT
I usually do it by finding out the smtp address to the cellualr subscribers sms address, and send the message to that address. To find out an email address that ends up in ones sms inbox: send an email from the phone to any other email address using sms (most american phones allow you to send emails using sms), look at the from field. Verizon is: [EMAIL PROTECTED] Sprint is: [EMAIL PROTECTED] wher phonenumber is a ten digit phone number. I'm not sure about cingular, att, and nextel. On Wed, 5 Jan 2005 09:54:00 -0600, Rich Adamson [EMAIL PROTECTED] wrote: We all mostly know that * as well as various SIP phones support SMS. While the final setup is somewhat of a mystery, there are reports of those lucky souls who have it working. We also know that in order to send an SMS to a mobile phone, we need to connect to some SMS message center and get the word out that way. Now, here's the new (?) element: How can I *accept* messages on my voip-based US landline? I know that if I send an SMS from my T-Mobile phone to a friend's Verizon phone, the message goes through, so somewhere there must exist a national message center that knows which carrier to hand the message off to. Technically it should be possible to register a phone number with them to receive messages sent from cell-phones or from other * systems, and then to receive these messages through * and onto a SMS capable IP phone...? Who knows more about this? Based on previous postings, the SMS thingie is primarly a european thing and is rather different from the US cellular implementation. Since you mentioned T-Mobile, I'm assuming you're in the US. If that assumption is correct, then its not likely you're going to be able to accomplish your objective without implementing some sort of site-specific role-your-own mechanism (eg, I don't know of any US cellular company that would sell you a sms address for your pbx). ___ 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] Sending DTMF to PSTN problem with SIP
You can configure the gain to be lower on the SPA2000 via the web interface - Ido not remember the exact location, but you will find it under advanced settings. --- CClarke [EMAIL PROTECTED] wrote: Dear All ~ I have * setup running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the SIP2000 to use inband dtmfmode (as opposed to auto), and likewise in sip.conf, but no success. btw, I've also set relaxdtmf=yes in zapata.conf since inbound calls sometimes seem to have trouble dialling extensions. A soft IAX phone (e.g. DIAXPhone) works ok, so I suspect my SIP2000/sip.conf setup, but can't see what I'm doing wrong. Christina. ___ 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 __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Digium T100P T1 Card
On Wed, 2005-01-05 at 10:23 -0700, Wiley Siler wrote: LOL - Thanks for not getting mad about my email. I just felt a little stung for being uneducated about T1s but we have to learn somewhere! I completely understand your concerns and will try to comply as best as I can. Again, thanks for being such a contributor to the this support system!! Until something is said that is blatantly slap in the face nasty, don't assume that a comment is meant to berate. For my one comment about the amount of knowledge left to learn, you should have accepted it as a measuring stick commenting on the depth of knowledge required to get to where you wanted to be and the estimated amount you already possessed. So while it might have been a bit of criticism, I went ahead and was willing to lay out examples that matched your situation to help fill in the gaps. Consider that a sign I felt you were worth me spending my time on and your likely hood of understanding the information I was about to give you. To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too mcuh to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. I don't know your area, and I don't think it has been mentioned. It might be a good idea to look into what it costs to break the contract, get DSL installed and your voice lines as a fractional T1 or PRI. DSL is usually quite a bit more inexpensive than a fractional T1 but at the cost of a reduced priority if you have a line failure. A full data T1 in my area seems to run about $750 a month, but I can get a business class DSL 3meg for $85. As you can see, it wouldn't take long for the difference in service charges to add up to the cost of breaking the contract. You then can look at what it will cost to get a telco to drop a T1 into your office space. Last quote we where involved with was around $200 for the loop and then whatever service you wanted on it. So 12 lines would probably run around $500 or so. Compare that to your service now and see what you think. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. It is a shame you already bought the equipment as you may find that you want more than one T1 port. But working within your constraints now, lets look at what can be done and what needs to be available as a feature. Obviously you need some analog FXS ports for your fax machines. With only 1 T1 span available, you probably need to follow the suggestion I made before about passing the T1 through a channel bank and using the spare channels to signal back to the FXS or FXO ports in the channel bank. Remember that your T1 interface can have 24 channels and if you only have 12 phone lines being passed from one external to the channel bank interface(PSTN side) to another external to the channel bank interface(T100P side), you can use the remaining 12 channels in the T100P side to signal back to analog ports on the channel bank. Granted this doesn't let you use PRI. That is why I suggested you look at EM wink. You still get your DIDs but they can be passed from one machine to the next without much trouble. What do you plan to use for phones in your office? SIP or analog? If analog, you will definately want another T100P card so you can bring the T1 line in directly to the first T100P card and then use the second to connect all 24 channels of the second T100P card to a channel bank. Or how about, you explain a bit more about where you want to go instead of where you are at so the end point can be planned then you can decide how to get there. I will have to google up ILEC and CLEC for more info b/c that is new to me as well. Single quote is me(Steven) from the previous message come from a ILEC(former baby bell) or a CLEC(competes with ILEC). Your ILEC is the Incumbent Local Exchange Carrier. Or what many used to call the Baby Bells, what was left after the
Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
On Tue, 2005-01-04 at 17:05 +1100, Eric Bishop wrote: And I thought it was just me going crazy. I have the exact same issue on a HP-Compaq DL360 G4 server (1U rackmount version). I have tried everything that has been mentioned here and more. Even replaced the TE410P card (so know it's not the card). I have tried with FC2, FC3 and RHEL 3. Have tried kernel 2.4.X and 2.6.X. Have tried vanilla kernels and stock fedora kernels. Have tried every known BIOS tweak. I have tried a different card in the slot (firewire card) and that does show interrupts, but no matter what I do I can't get the TE410P to show any interrupts. Loading the zaptel driver appears to work but the lights on the TE410P just go off (rather than the normal blinking). My /proc/interrupts always looks as follows: If anyone has the solution to this I owe you big! [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 0: 111005341IO-APIC-edge timer 1: 9IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 66IO-APIC-edge i8042 14: 7870IO-APIC-edge ide0 185: 0 IO-APIC-level t4xxp 193: 26141 IO-APIC-level cciss0 201:1139611 IO-APIC-level eth0 NMI: 0 LOC: 111010062 ERR: 0 MIS: 0 [EMAIL PROTECTED] ~]# Have also tried replacing the card, changinf PCI slots and messing with the BIOS all with the same result. If anyone can help I would be very grateful.. While browsing the Wiki, I found this bit of information. Maybe it will help you out some. http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Music from Freeplay music included in * ??
In article [EMAIL PROTECTED], John Middleton [EMAIL PROTECTED] wrote: Hi, Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk from the digium web server - Whats the CVS command for a 'head' install ? They should be in /usr/src/asterisk/sounds, but they don't appear to have the v1-0_stable tag: # cd /usr/src/asterisk/sounds # cvs log fpm-calm-river.mp3 RCS file: /usr/cvsroot/asterisk/sounds/fpm-calm-river.mp3,v Working file: fpm-calm-river.mp3 head: 1.2 branch: locks: strict access list: symbolic names: v1-0-2: 1.2 v1-0: 1.2.0.2 v1-0-1: 1.2 v1-0-0: 1.1 v-1_0_RC2: 1.1 keyword substitution: kv total revisions: 2; selected revisions: 2 description: revision 1.2 date: 2004/09/27 20:03:59; author: markster; state: Exp; lines: +1 -1 Strip mp3 id3 headers (bug #2525) revision 1.1 date: 2004/08/01 14:19:04; author: markster; state: Exp; Rename newp to newpvt (bug #2190), change hold music. = Try checking out (or updating) with just v1-0 instead. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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: lcdproc and asterisk
Corvin wrote: Matt Riddell wrote: Corvin wrote: Hi! I would like to use lcdproc and asterisk. Any hints or links? Maybe someone has experience in such matter. I am working on such solution. I've heard of SAPBX. Thanks for any help. Hi, I was working with someone on this until my BB Forum fell over. Drop me a line if you need any help/have any questions regarding this. Hi, This is my first da on this project. I haven't it stared. Now I am preparing LCD and some documentation. So *any* help hint, program highly appreciate :). Corvin Sorry, I don't have a program for Linux any more - I could have another one up and running in a few days. So, recommendations: Get a cheap hd4480 controller based LCD, wire it up using the WinAMP wiring plan (pretty simply, but soldering can be a bit messy). Install lcdD (the server), set it up for winamp wiring/buttons - if you have them. Write an AGI script to send information to the LCD (we were working on AstLCDd - a simple wrapper for voicemail info etc, but would maybe do it differently now). You haven't actually mentioned what you hope to achieve... :-) Are you wanting stats/voicemail etc, or do you want the buttons to perform various actions on your server I.E. reload, change IP address etc. It's probably best if we take this off-list as it is only minimally associated with Asterisk, and the first things you will have to do will be dealing solely with LCDproc (just send me an email to [EMAIL PROTECTED]) Also, a note to the guy in Europe who I was working on AstLCDd with - drop me an email and I will set up a new forum - it seems more and more people are interested in this. -- Cheers, Matt Riddell ___ Daily Asterisk News: http://www.sineapps.com/news.php for html http://www.sineapps.com/rssfeed.php for rss ___ 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] Happy Wednesday Morning SMS question, slightly OT
Some commerical SMS gateways can provision a # for routing inbound messages. An example or 2 would be clickatell and ippipi ___ 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] Happy Wednesday Morning SMS question, slightly OT
That's a known, yet not feasible work-around over accessing an SMS-center directly. But the question remains how to accept IMCOMING messages with *. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 05, 2005 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Happy Wednesday Morning SMS question, slightly OT I usually do it by finding out the smtp address to the cellualr subscribers sms address, and send the message to that address. To find out an email address that ends up in ones sms inbox: send an email from the phone to any other email address using sms (most american phones allow you to send emails using sms), look at the from field. Verizon is: [EMAIL PROTECTED] Sprint is: [EMAIL PROTECTED] wher phonenumber is a ten digit phone number. I'm not sure about cingular, att, and nextel. ___ 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] CallerID in Australia Analogue PSTN Phone
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
Sure was. It was me calling myself from my mobile (cell) phone, and
that definitely has CLID enabled. In AU CLID is enabled by default.
Do you know if the Digium X101P has problems with reading CLID on the
line? There is a wiki that says that in AU the DEFAULT_CIDRINGS needs
to be =2 rather than the default =1 and I have set that; perhaps I
should reverse that and try again.
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RE: [Asterisk-Users] Digium T100P T1 Card
Good points all. Apologies and thanks again. I guess I am the master at leaving out pertinent information. We are locate in Phoenix AZ. I currently have a fully functional phone system built on * that uses Polycom IP 500s over SIP internally. Lines from the AdtTran are delivered via two TDM400P cards in my Aasterisk box. Both the box and the client phones sit behind my Cisco firewall. I am only servicing 12 extensions internally and a single fax machine. Growth is is expected to only increase to 20 SIP devices/users in the next 6-9 months. The goal is to get better line quality, have DIDs, and increase line count. How we get there only has to follow two parameters. It has to be cheap cuz the boss is... It has to work because the boss wants perfection for the lowest dollar... I am sure you can imagine. 8) That is a ton of options. Thanks! Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, January 05, 2005 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Digium T100P T1 Card On Wed, 2005-01-05 at 10:23 -0700, Wiley Siler wrote: LOL - Thanks for not getting mad about my email. I just felt a little stung for being uneducated about T1s but we have to learn somewhere! I completely understand your concerns and will try to comply as best as I can. Again, thanks for being such a contributor to the this support system!! Until something is said that is blatantly slap in the face nasty, don't assume that a comment is meant to berate. For my one comment about the amount of knowledge left to learn, you should have accepted it as a measuring stick commenting on the depth of knowledge required to get to where you wanted to be and the estimated amount you already possessed. So while it might have been a bit of criticism, I went ahead and was willing to lay out examples that matched your situation to help fill in the gaps. Consider that a sign I felt you were worth me spending my time on and your likely hood of understanding the information I was about to give you. To further explain my siutation, I should give you some more background on my setup. My current setup has an AdTran 616 on the wall breaking out my 6 analog lines and delivering my data to the office. I have two TDM400P cards receiving 6 analog lines which are used for both fax and voice. I have had numerous problems with this ISP and I just want to get away as soon as possible. Problem is, I have a contract that won't expire for a while so I need to use these lines for something. The ISP wants a contract extension and some setious cash to do the upgrade. Better to just seek alternate service. I originally bought my T100P thinking I would get digital lines and all the goodies involved. Then budget constraints and an ISP that wants too mcuh to convert me to Digital lead to a temporary solution. I would use the analog lines for a while longer. Well, that has run its course and I have to get to something more stable. The PRI card looks pretty good at this point. I don't know your area, and I don't think it has been mentioned. It might be a good idea to look into what it costs to break the contract, get DSL installed and your voice lines as a fractional T1 or PRI. DSL is usually quite a bit more inexpensive than a fractional T1 but at the cost of a reduced priority if you have a line failure. A full data T1 in my area seems to run about $750 a month, but I can get a business class DSL 3meg for $85. As you can see, it wouldn't take long for the difference in service charges to add up to the cost of breaking the contract. You then can look at what it will cost to get a telco to drop a T1 into your office space. Last quote we where involved with was around $200 for the loop and then whatever service you wanted on it. So 12 lines would probably run around $500 or so. Compare that to your service now and see what you think. So getting back to the T1 PRI issue (and I am playing catch up here), my goal is to just deliver new service into this office over my T100P and just dump nothing but fax out those old lines. That way I can reserve the digitals for our truly important calls and still reap the benefit of having those old analog lines. It is a shame you already bought the equipment as you may find that you want more than one T1 port. But working within your constraints now, lets look at what can be done and what needs to be available as a feature. Obviously you need some analog FXS ports for your fax machines. With only 1 T1 span available, you probably need to follow the suggestion I made before about passing the T1 through a channel bank and using the spare channels to signal back to the FXS or FXO ports in the channel bank. Remember that your T1 interface can have 24 channels and if you only have 12 phone lines being passed from one external to the
[Asterisk-Users] chan_oh323 Module for Asterisk
If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: Wednesday, January 05, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_oh323 gatekeeper Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Last callers script?
That sounds like it might just be the ticket Roger. I like the web page idea too. Would you be willing to share it please? Thanks Mike On Wed, 05 Jan 2005 11:32:08 -0500, Roger Gulbranson [EMAIL PROTECTED] wrote: On Wed, 2005-01-05 at 11:00, Mike Dent wrote: Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? I have an AGI script (a modified version of calleridnamelookup.agi) that, among other things, stores the channel and callerid in a mysql DB. The AGI is called from within my IVR processing on all the inbound channels. I happen to use this for a web page that displays the most recent 20 calls. Writing an AGI script to take a channel and find the last inbound callerid should be an easy thing to do (once you have the data). No doubt there are other ways to achieve the same result. DBget/DBput could be used, for example. ___ 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] Happy Wednesday Morning SMS question, slightly OT
From: Rich Adamson [mailto:[EMAIL PROTECTED] implementation. Since you mentioned T-Mobile, I'm assuming you're in the US. The phrase voip-based US landline should have given that away as well :) On a related note, T-Mobile or T-Mobil is the European parent of T-Mobile US (formerly VoiceStream) going to be able to accomplish your objective without implementing some sort of site-specific role-your-own mechanism (eg, I don't know of any US cellular company that The question was partially a how does this work? and can I do that? type question. Somehow the providers currently work together to make message delivery nearly seamless, even with GSM phones in other countries. Would be interesting to know what happens with an SMS that is sent to a landline. ___ 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] Vonage WiFI Phone...
On Tue Jan 04, 2005 at 01:27:21PM -0500, Philippe Daoust wrote: Anybody know anything about this F-1000 phone? 100 hours of battery life, not bad at all... http://www.utstar.com/Solutions/Document_Library/Handsets/docs/WiFi/F1000DataSheet.pdf This quotes 48-80 hours standby, so you can probably reckon on it being towards the lower end of that in reality. Would be interested to hear the retail price of these (rather than the Vonage bundled price) Simon -- Simon Lockhart | * Sun Server Colocation * ADSL * Domain Registration * Director|* Domain Web Hosting * Internet Consultancy * Bogons Ltd | * http://www.bogons.net/ * Email: [EMAIL PROTECTED] * ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP Phone suggestion.
Does any body know an IP phone that has at least 2 line appearances, POE, is around $150 USD, and works nice with *. I've been looking at the UIP 200 but it's only a single line phone, and I'm looking for something that has at least 2. ___ 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] X-lIte behind NAT and Asterisk behind NAT
Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the firewall rules are correct, because I have also open up port 21, and he can successfully ssh into my Asterisk box. Any ideas/pointers? Thanks in advance Richard ___ 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] X-lIte behind NAT and Asterisk behind NAT
SSH runs on port 22, so either that's a typo or you've got something else going on. Did you forward port 5060, or just open it on the router? You probably need to forward it to the Asterisk box's IP. __ Dana Olson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of richard Sent: Wednesday, January 05, 2005 4:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] X-lIte behind NAT and Asterisk behind NAT Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the firewall rules are correct, because I have also open up port 21, and he can successfully ssh into my Asterisk box. Any ideas/pointers? Thanks in advance Richard ___ 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 Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this message in error, please contact the sender and delete the material from any system. ___ 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] www.cuphone.com PCI hardware
Hi guys, I've just started playing with Asterisk, and I must say that I'm very impressed. I'm now looking at hooking this up to a single phone line, and looking for cheapish hardware to do so. While doing this, I've stumbled across a Personal Phone Gateway PCI card at: http://www.cuphone.com/products/ppg/index.htm Does anyone know if this works as an FXO? and if so, does Asterisk support it? I can't really see much on the web about it. I have heard rumour about a modified motorola chipset modem that can work as an FXO, and I'm wondering if this is it... Ideas anyone? -- Signed, Steven Haigh I am root. If you see me laughing, you'd better have a backup. ___ 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] funny little question regarding asterisk as a pbx vs a key system [slightly OT]
To all those who answer 50+% of the questions on this list with '* cannot do that since it is a pbx and what you talk about is the functionality that a key system provides'... I pose a question. What would it take - from any point of view you wish to use - to change that statement to '* cannot do that because it is a pbx, but [insert name of key system version of * here] can, goto www.[insert domain here].com and download it'? I pose this question simply as a personal point of interest. If the idea is feasible, I would like the think a team could be put together to produce such a beast - maybe even integrating it into *... Anyway - more ramblings from the mumbling idiot. :) -Chris ___ 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] RFI: Creating a database of DID providers
What type of information would you be looking for from the DID providers? I know the company that I am with works with one DID provider, but might be interested in expanding beyond that. I would also recommend/request that the database have info in it letting people know about the outbound call providers. I am not sure if that is a common idea to have an * server running outbound calling lines to many different phone markets, let alone 4+ area codes, but it is something I am involved with. Dan On Wed, 29 Dec 2004, Paul Crick wrote: atus: RO X-UIDL: B0069077940.MSG Cross posted from asterisk-biz: Is anyone willing to host/manage a website that people can simply browse that lists all current DID providers and their coverage areas? It's a good idea and probably not too hard to implement, it's just a case of deciding how far you want to go.. are areacodes good enough? or do you need to go to NPA-NXX level and start talking about rate centers etc? Ok.. I'm going to have a stab at this.. I'd like to have some kind of search mechanism similar to that at www.voipreview.org where you can select country and area (by state/city? or would people prefer by areacode?) then generate a list of all providers than can supply DIDs in that area, together with setup/rental charges, per minute charges, etc. Before I go reinvent the wheel totally from scratch, is there anyone out there that has data in electronic form that they use already for this sort of thing? I'm looking for country code listings, area code listings, NPA-NXX to city name listings etc. Replies to the list, or forward data files to web-dids at ivrl.com Cheers Paul ___ 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. ___ 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] Agent login state saving?
On Fri, 31 Dec 2004 [EMAIL PROTECTED] wrote: From configs/queues.conf.sample: [general] ... ; Persistent Members ;Store each dynamic agent in each queue in the astdb so that ;when asterisk is restarted, each agent will be automatically ;readded into their recorded queues. Default is 'yes'. Looks like this is only in cvs-head. Are you using that in production? AFAIK, there have been some serious changes to the ways queues work in cvs-head. -- Jon Lewis | I route Senior Network Engineer | therefore you are Atlantic Net| _ http://www.lewis.org/~jlewis/pgp for PGP public key_ ___ 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] Happy Wednesday Morning SMS question, slightly OT
Not all providers bind the number to a email address. I havent set it up, but in terms of sms, if asterisk could send out the message to a URL, or connect using SMPP then it could be done. Asterisk --- over http ---url--- url parses number in the GET request and then fires that request by a provider to the number long/short Or u could set up kannel on a another box, connect kannel to a provider, it just like we connect to voip providers, we can also get accounts with sms providers, then asterisk sends message to port on kannel machine, kannel will then send out via the SMS provider the message. Iqbal On 1/5/2005, C F [EMAIL PROTECTED] wrote: I usually do it by finding out the smtp address to the cellualr subscribers sms address, and send the message to that address. To find out an email address that ends up in ones sms inbox: send an email from the phone to any other email address using sms (most american phones allow you to send emails using sms), look at the from field. Verizon is: [EMAIL PROTECTED] Sprint is: [EMAIL PROTECTED] wher phonenumber is a ten digit phone number. I'm not sure about cingular, att, and nextel. On Wed, 5 Jan 2005 09:54:00 -0600, Rich Adamson [EMAIL PROTECTED] wrote: We all mostly know that * as well as various SIP phones support SMS. While the final setup is somewhat of a mystery, there are reports of those lucky souls who have it working. We also know that in order to send an SMS to a mobile phone, we need to connect to some SMS message center and get the word out that way. Now, here's the new (?) element: How can I *accept* messages on my voip-based US landline? I know that if I send an SMS from my T-Mobile phone to a friend's Verizon phone, the message goes through, so somewhere there must exist a national message center that knows which carrier to hand the message off to. Technically it should be possible to register a phone number with them to receive messages sent from cell-phones or from other * systems, and then to receive these messages through * and onto a SMS capable IP phone...? Who knows more about this? Based on previous postings, the SMS thingie is primarly a european thing and is rather different from the US cellular implementation. Since you mentioned T-Mobile, I'm assuming you're in the US. If that assumption is correct, then its not likely you're going to be able to accomplish your objective without implementing some sort of site-specific role-your-own mechanism (eg, I don't know of any US cellular company that would sell you a sms address for your pbx). ___ 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] voiptalk.org IAX service - user experiences
Paul wrote: Having all sorts of nightmares getting IAX working from voiptalk.org Originally I did too but it was all my fault. We have been using VoIP Talk for about 3 months and have no complaints. Getting outbound IAX (from PBX to PSTN via VoIPTalk) is straight forward the guide on their web site is accurate. Provided you have bought IAX credits you should be able to use IAX successfully. To receive calls via IAX you must have a number (either a free 0870 number or a paid for geographical). You must also ask VoIPTalk support to add the IP address of your * server to the telephone number. This IP address can be the address of your gateway if you are using NAT. Bill Seddon ___ 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] Allowing pooling or rollover for inbound callson VoicePulse
This is more of a VoicePulse thing than an asterisk thing - you'd need them to roll over to 5551001 after presenting you with 4 calls on 5551000.. although really, this is kinda silly.. It would be better to talk to them about upping the limit on the number of simultaneous calls you can receive per DID, no? Paul ___ 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] chan_oh323 Module for Asterisk
I got it, but email it to the list is not a good option. Who 're interested just email me, I'll send it asap. But AFAIK, you still need the wrapper. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, January 05, 2005 5:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] chan_oh323 Module for Asterisk If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Humberto Aicardi Sent: Wednesday, January 05, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_oh323 gatekeeper Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] IP Phone suggestion.
Does any body know an IP phone that has at least 2 line appearances, POE, is around $150 USD, and works nice with *. I've been looking at the UIP 200 but it's only a single line phone, and I'm looking for something that has at least 2. This information is on the wiki... www.voip-info.org. The closest you'll find is a Polycom SoundPoint IP300 with the POE adapter cable, around $175. Two line appearances, but listen-only speakerphone. ___ 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] Vonage WiFI Phone...
On Tue Jan 04, 2005 at 01:27:21PM -0500, Philippe Daoust wrote: Anybody know anything about this F-1000 phone? 100 hours of battery life, not bad at all... The peanut gallery chimed in on this yesterday: http://slashdot.org/article.pl?sid=05/01/04/1816228tid=193tid=215 ___ 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] Twin Cities Asterisk meeting this Saturday?
I saw the post on the wiki a last month stating the meeting was this Saturday. Is that confirmed? Still on for 1/8? Roger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debug channel n
I can't get the debug channel command to work. In each case * responds with No such channel I've tried: debug channel 1 debug channel Zap/1 debug channel Zap/1-1 debug channel 25 debug channel Zap/25 debug channel Zap/25-1 etc. The zap show channels command shows all channels to be present. Am I using the correct syntax? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-lIte behind NAT and Asterisk behind NAT
Hi, First of all you have to configure the externip and localnet parameters at the sip.conf file. You have to write the external ip address of your internet connection to the extern ip parameter like exterip=XXX.YYY.ZZZ.WWW and your local address for ex.localnet=192.168.1.0/255.255.255.0 after all you have to configure your X-lite's network parameters to use your clients external ip address for your SIP communication. After all it will be working if you have further problems you can read the documents at the http://www.voip-info.org site by searching SIP NAT Yusuf Alakavuk Teknik Danisman - Technical Consultant Grid Bilisim Teknolojileri A.S. Kustepe Mahallesi Leylak Sokak Murat Is Merkezi A Blok Kat:2 Daire:9 34387 Sisli Istanbul Türkiye Tel : +90 (212) 336 92 55 Fax : +90 (212) 266 25 50 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of richard Sent: 05 Ocak 2005 Çarsamba 23:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] X-lIte behind NAT and Asterisk behind NAT Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the firewall rules are correct, because I have also open up port 21, and he can successfully ssh into my Asterisk box. Any ideas/pointers? Thanks in advance Richard ___ 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] chan_oh323 Module for Asterisk
That's the problem. You need the chan_oh323.so and the oh323wrapper. You can try it, but, i guess it i'll not work. A little help from Michael Manousos at this point i'll be great ;) Tomorrow i'll try to get it working, but, if i can't, maybe i'll need to do downgrade asterisk chan_oh323 versions. Joo Amaro Tenorio, Leandro wrote: I got it, but email it to the list is not a good option. Who 're interested just email me, I'll send it asap. But AFAIK, you still need the wrapper. LTenorio -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, January 05, 2005 5:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] chan_oh323 Module for Asterisk If anyone in the list has a working version of the chan_oh323.so file for Fedora Core 2 and Redhat, can he email the same to the list as attachment. This will reduce the pain for many of the users who are trying to compile the same from the libraries, which never seemed to work. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Humberto Aicardi Sent: Wednesday, January 05, 2005 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] chan_oh323 gatekeeper Hi folks, Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? Thanks, Humberto Aicardi NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Last callers script?
On Wed, 2005-01-05 at 15:52, Mike Dent wrote: That sounds like it might just be the ticket Roger. I like the web page idea too. Would you be willing to share it please? I've attached the agi script. My web site is written in Mason which probably doesn't interest many folks. The table I use is: mysql describe asterisk_callerid_history; +---+-+--+-+-+---+ | Field | Type| Null | Key | Default | Extra | +---+-+--+-+-+---+ | timestamp | datetime| YES | MUL | NULL| | | callerid | varchar(80) | YES | | NULL| | | channel | varchar(50) | YES | | NULL| | +---+-+--+-+-+---+ If you know SQL figuring out how to 'select' the table for either a web page or a lookup agi script should not be a big deal. calleridnamelookup.agi Description: Perl program ___ 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] X-lIte behind NAT and Asterisk behind NAT
On Wed, 2005-01-05 at 16:03, richard wrote: Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the firewall rules are correct, because I have also open up port 21, and he can successfully ssh into my Asterisk box. Any ideas/pointers? Thanks in advance Richard ___ 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 Isn't ssh on port 22? Dave ___ 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] IP Phone suggestion.
-Original Message- Does any body know an IP phone that has at least 2 line appearances, POE, is around $150 USD, and works nice with *. I've been looking at the UIP 200 but it's only a single line phone, and I'm looking for something that has at least 2. --- Yes, Polycom 300, But it does not have a speaker phone NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] sip.conf asterisk to vonage
i have tried to connect my asterisk server to vonage like this: Sip.conf: register = 1yournumber:secret@atlas-east.vonage.net:5060 [vonage] type=friend username=1yournumber secret=secret host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1yournumber fromdomain=atlas-east.vonage.net canreinvite=no nat=yes context=default Extensions.conf: [default] exten = _1yournumber,1,Dial(SIP/111) and i tried port 5061 also instead of 5060 with no luck. When i look at the log messages from the CLI i get the message han_sip.c:3986 sip_reg_timeout: Registration for '1yournumber@atlas8.atlas.vonage.net' timed out, trying again Any clues? mohammed ___ 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] debug channel n
I can't get the debug channel command to work. In each case * responds with No such channel I've tried: debug channel 1 debug channel Zap/1 debug channel Zap/1-1 debug channel 25 debug channel Zap/25 debug channel Zap/25-1 I beleive you have to have the channels in use (as per show channels) AND on zap devices include the span number eg Zap/1-1 that how it works here ___ 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 / * re-register issues
After about 4 or 5 minutes, however, I cannot get incoming calls. It either just rings or goes busy, and never executes the dialplan in extensions.conf. Broadvoice has four servers that may send your * server calls. This was my sip.conf setup until last week (when I cancelled Broadvoice): register = phonenumber:[EMAIL PROTECTED]/phonenumber [bv-out] type=peer username=phonenumber fromuser=phonenumber secret=password host=sip.broadvoice.com fromdomain=sip.broadvoice.com disallow=all allow=ulaw dtmfmode=inband canreinvite=no [bv-in-1] type=friend host=147.135.8.128 context=from-bv dtmfmode=inband canreinvite=no [bv-in-2] type=friend host=147.135.0.128 context=from-bv dtmfmode=inband canreinvite=no [bv-in-3] type=friend host=147.135.4.128 context=from-bv dtmfmode=inband canreinvite=no -- Nabeel Jafferali tel: 416.491.9136 (toronto) 646.225.7426 (new york) fwd: 46990 email/msn : nabeelatjafferali.net ___ 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] Digium T100P T1 Card
On Wed, 5 Jan 2005, Wiley Siler wrote: A consultant so I can get a T1 PRI on my wall and use it with my Asterisk box? LMAO. That is the dumbest thing I have ever heard. I need a consultant so I can get a T1 with PRI? Please. I am just trying to better understand how the Digium PRI card works and how it interconnects to the ISP. The telecom system is really a lot more complex than most people think. Getting someone who knows the field is really a good idea for almost anything more complex than a single analoge connection. If you do know the stuff yourself, great. Otherwise your time will have to be pretty cheap to compensate for the time it takes to know enough to build an isdn pbx with voip. Peter ___ 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] chan_oh323 gatekeeper
Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? I have this in my extensions.conf. The oh323.conf has gatekeeper disabled and nothing else specific to the 192.168.99.83. Works just fine to place calls to a Cisco fxo gateway. exten = s,3,Dial(OH323/[EMAIL PROTECTED]) ; H.323 Protocol Adi ___ 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] Realtime
Hi, Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such switch 'Realtime' What does this message mean ? Something wrong with the switch statement in my extensions.conf or maybe is the module net correctly installed ? Thnx. ___ 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] Streaming Audio - Music On Hold Feature
I was wondering, does anyone know if it is possible to have a stream of audio coming from a Microsoft compressed audio stream fed to the caller if they are placed on hold and if so how might this be done? Dan - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues - announcements and not busy members
Hi! I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. Is this possible to change so the announcement will be played even if there are free member-extensions? I think that would be nice (well it's not how ACD-groups usually works but anyway) Best regards, Lars --- http://www.fredriksson.net/ mailto:lars at fredriksson dot net ___ 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] Aaargh Gentoo updated some packages now * won't start
After emerging some updates this morning asterisk 1.0.3 fails to start I get the following errors: ..Jan 6 00:39:24 WARNING[28998]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Jan 6 00:39:24 ERROR[28998]: chan_zap.c:6197 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 6 00:39:24 ERROR[28998]: chan_zap.c:9141 setup_zap: Unable to register channel '1' Jan 6 00:39:24 WARNING[28998]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Jan 6 00:39:24 WARNING[28998]: loader.c:440 load_modules: Loading module chan_zap.so failed! root # Ouch ... error while writing audio data: : Broken pipe I already re-emerged asterisk and zaptel but it's still not working. Ideas anyone? ___ 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] Digium T100P T1 Card
Well, I wont say my time is cheap or that I know everything about the T1s. However, I did manage to build my PBX on *, implement Polycom IP 500 phones pulling configs from the network, and script my own extensions all with an initially minimal understanding of Linux. I have dealt with problems as they arose, sought out solutions on the wiki and elsewhere, learned more about Linux and I am very comfortable with the system now. Telecom is complex but that does not mean that only a contractor can get it done. If I had money to pay a contractor, I would probably have had money to buy a boxed PBX in the first place. Cost effective has always been the greatest selling point to me regarding my * PBX. Just enjoying the challenge and learning new things has been good too. I have no question that I am capable of implementing this. How long it will take and how hard are of course different questions. That I have only myself as a resource is not. After reading the wiki on setting up the PRI it does not look that complicated to me. The recompile is the only portion that looks time consuming. That may prove wrong or right. Regardless, I will be going the course alone... And that's OK. 8) Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Wednesday, January 05, 2005 4:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Digium T100P T1 Card On Wed, 5 Jan 2005, Wiley Siler wrote: A consultant so I can get a T1 PRI on my wall and use it with my Asterisk box? LMAO. That is the dumbest thing I have ever heard. I need a consultant so I can get a T1 with PRI? Please. I am just trying to better understand how the Digium PRI card works and how it interconnects to the ISP. The telecom system is really a lot more complex than most people think. Getting someone who knows the field is really a good idea for almost anything more complex than a single analoge connection. If you do know the stuff yourself, great. Otherwise your time will have to be pretty cheap to compensate for the time it takes to know enough to build an isdn pbx with voip. Peter ___ 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] Read() timeout hangs up the line
Hi list,
I am having some difficulty implementing a certain dialplan where the
following
happens. If the first Dial() is not answered, I want to play a small
greeting then
ask the caller to either hold the line (try calling again) or press 1
to leave
voicemail.
exten = s,1,Dial(${BLAH},10,Tt) ; Dial 10 sec
exten = s,2,Answer
exten = s,3,Playback(greeting)
exten = s,4,Playback(werebusy)
exten = s,5,DigitTimeout(1)
exten = s,6,ResponseTimeout(3)
exten = s,7,Read(WHAT,holdormsg,1) ; Hold the line, or
press 1 to leave a msg..
exten = s,8,Gotoif($[${WHAT} = 1]?30)
exten = s,9,Dial(${BLAH},15,Ttm) ; Dial another
15 sec with music on hold
exten = s,10,Goto(7) ; Loop
My problem is that if the caller doesn't press a key when prompted, and
the Read()
is allowed to time out (3 seconds), (s,7) returns non-zero and asterisk
hangs up
on the caller without further execution. I want it to continue down the
priorities and
redial the line, with hold music.. It doesn't even get to test the
value of ${WHAT}
if nothing is entered. However, if the caller enters an number other
than 1, it will
perform properly and redial the line.
-- Executing Read(vpb/1-1, WHAT|holdormsg|1) in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'holdormsg' (language 'en')
-- User entered ''
== Spawn extension (blah, s, 9) exited non-zero on 'vpb/1-1'
== vpb/1-1: Hangup requested
== vpb/1-1: Ending record mode (1/yes)
== vpb/1-1: Ending play mode on vpb/1-1
== vpb/1-1: Hangup complete
Any ideas what could be wrong ?
Cheers
Troy
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Re: [Asterisk-Users] Realtime
Serge Schumacher wrote: Hi, Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such switch 'Realtime' What does this message mean ? Something wrong with the switch statement in my extensions.conf or maybe is the module net correctly installed ? Perhaps you might consider posting your extensions.conf and other relevent troubleshooting details? ___ 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] TE410P card in an HP-Compaq DL380 G4 server
Well it's clear now that this is not an isolated issue. Has anyone been in touch with Digium about this issue? I have logged a support issue with them, but thus far have not received a response. Anyone had better luck with Digium support and the Compaq/HP G4 server series? On Wed, 5 Jan 2005 18:05:22 +0100, Tais M. Hansen [EMAIL PROTECTED] wrote: On Monday 03 January 2005 19:34, [EMAIL PROTECTED] wrote: Has anyone had success using a TE410P card in an HP-Compaq DL380 G4 server? We're struggeling with the same thing right now. We have several TE410Ps working on DL380G3s, but have so far been unsuccessful in getting it to work on the G4. Our G4 config is dual xeon 3.6ghz, 2gb ram, kernel 2.6.10 and 2.4.28. zaptel and wct4xxp modules loads fine. At this point the flashing red lights on the wct4xxp are turned off. zttool shows all spans are OK, no matter if there are anything plugged in. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 ___ 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] TDM04B vs Dell
Hi all, I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. TIA, Michael Swan Neon Software, Inc. ___ 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] Out the box solutions?
Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind around. It's a completely different landscape! And why is it that /etc/modules.conf in one release is /etc/modprobe.conf in another? Is this beast not based on standards? But I digress. I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did. At least with unix I was able to get a dial tone! Not so much with this flavor of linux. Each time I run modprobe wcfxs I get the following errors in /var/log/messages: Jan 5 17:57:59 asterisk wait_for_sysfs[2782]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2784]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap2' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2786]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap3' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2788]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap4' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] I'm not so interested in notifying these guys at lists.sourceforge.net, since I'm only interested in running asterisk. Once I commit to actually using linux I might participate in their forum, but not yet :) So ... the question: What flavor of linux does asterisk actually run on Out the box? I'm not scared to compile asterisk, but I'm not at all interested in recompiling a linux kernel. Of course if that is the only way, then I guess I'll just bite another bullet. Hell! I want this PBX to work so bad that I can almost taste it! Please advise. lane ___ 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] Allowing pooling or rollover for inbound calls on VoicePulse
On Wed, 5 Jan 2005 15:12:08 -0500, Zeno Lee wrote: My goal is to have only 1 primary phone number that can seamlessly pool multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to forward or roll over the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000 remains open for connections. Theoretically I would like to maintain 11 simultaneous connections all coming in from just the 212 555 1000 account. This situation applies if I wanted to voice conference more than 4 people on a single phone number (with multiple accounts on VoicePulse) Is this possible with Asterisk? I haven't used VPC in a while but I suspect what you ask is already in place. Not so much the rollover, which is really a hunt group. But VPC allows multiple simultaneous calls ongoing from one account. Back in the summer when I used them they allowed 6 calls on one account. You pay for all six, but they would come in on your primary number. You may not need the additional DIDs at all. I myself once conferenced 4 people using one VPC account. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
RES: [Asterisk-Users] chan_oh323 gatekeeper
You're right it works, but how about receiving calls, how can you register so the FXO gateways knows where to send the calls? Or I just setup the FXO gateway with the IP address of the * box? Humberto Aicardi -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Adi Linden Enviada em: Wednesday, January 05, 2005 9:28 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] chan_oh323 gatekeeper Until now I have used only SIP IAX2 with success and understand them pretty well. The point is that someone has asked me to configure an * box for them, the problem is that they want to use H.323. I have already compiled and tested the chan_oh323 with asterisk and works. The problem is that the tests need a gatekeeper, my question is: Do I need always need a gatekeeper? Or my FXO H.323 gateway can register with * ? I have this in my extensions.conf. The oh323.conf has gatekeeper disabled and nothing else specific to the 192.168.99.83. Works just fine to place calls to a Cisco fxo gateway. exten = s,3,Dial(OH323/[EMAIL PROTECTED]) ; H.323 Protocol Adi ___ 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] Aaargh Gentoo updated some packages now * won't start
Hi, Did you update the kernel or modutils? Maybe try to recompile Zaptel modules... bye, Samuel T. Cossette [EMAIL PROTECTED], 1.418.8o2.784o Well, that's for me to know and you to find out. Jeffrey, Blue Velvet After emerging some updates this morning asterisk 1.0.3 fails to start I get the following errors: ..Jan 6 00:39:24 WARNING[28998]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Jan 6 00:39:24 ERROR[28998]: chan_zap.c:6197 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 6 00:39:24 ERROR[28998]: chan_zap.c:9141 setup_zap: Unable to register channel '1' Jan 6 00:39:24 WARNING[28998]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 Jan 6 00:39:24 WARNING[28998]: loader.c:440 load_modules: Loading module chan_zap.so failed! root # Ouch ... error while writing audio data: : Broken pipe I already re-emerged asterisk and zaptel but it's still not working. Ideas anyone? ___ 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] Cisco 7920 and Asterisk - Anyone got this working?
I am having all sorts of probs. It just won't connect. Anyone got any example configs I could look at? Thanks Paul ___ 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] Out the box solutions?
Lane wrote: Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind around. It's a completely different landscape! And why is it that /etc/modules.conf in one release is /etc/modprobe.conf in another? Is this beast not based on standards? I think you mean conf.modules vs modules.conf, which are there for backwards compatibility. Yes, I concur they should stick with one or the other to not confuse newbies, but tell that to all the linux maintainers :) But I digress. I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did. There's your first problem :) Try Gentoo or Slackware. At least with unix I was able to get a dial tone! Not so much with this flavor of linux. Each time I run modprobe wcfxs I get the following errors in /var/log/messages: Did you read the docs, check the mailing lists and the wiki for information on kernel 2.6? you will have to issue 'make linux26' for that to work. Also, since the cvs release as of Nov 9th, it's now modprobe wctdm, not wcfxs So ... the question: What flavor of linux does asterisk actually run on Out the box? Many of them, but you will have to do some reading first. Or try one of the Asterisk Live cd's, or customized ISO installers. I'm not scared to compile asterisk, but I'm not at all interested in recompiling a linux kernel. You are going to have to get your hands dirty if you wish to accomplish anything productive. Or, hire a consultant. Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ 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] Polycom IP500 - problems with multiple simultaneous calls
Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the on-hold feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my dialplan go to the second appearance/registration. Instead, the second call rings on the first line, and allows you to put the first call on hold, and take the second call. To do so, though, you have to press the little down arrow and then press Answer. When the third call comes in, it will ring the 2nd line. I find this to be non-intuitive, but I can get used to it. My receptionists, however, are finding it REALLY painful. I'd just like to make the first call go to line appearance 1, the second simultaneous call to go to line appearance 2, etc. Maybe somebody figured out a neat dialplan thing to get this done. My config that doesn't do what I want looks like this: ; The first line appearance is registered to 18, the second to 1802, and the third to 1803 exten = 18,1,Dial(SIP/18,20) exten = 18,2,Voicemail(u18) exten = 18,102,Goto(1802,1) exten = 1802,1,Dial(SIP/1802,20) exten = 1802,2,Voicemail(b18) exten = 1802,102,Goto(1803,1) exten = 1803,1,Dial(SIP/1803,20) exten = 1803,2,Voicemail(b18) exten = 1803,102,Voicemail(b18) exten = 1803,103,Hangup I guess the phone just doesn't register as busy when there is only one call on a line. It has to have two calls on a line appearance to register as busy. Has anyone figured out how to disable this hold feature and just have the second call go to the second line, the third call to the third line, etc? Thanks, Noah Miller ___ 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] Out the box solutions?
A good alternative would be to try a free rebuild of RedHat Enterprise Linux, for example www.taolinux.org. Just use the 32 bit version, the 64 bit version (if you would have the cpu) gives me trouble compiling the kernel modules. With 32 bit Tao it runs almost out of the box and works like a charm. You get the (community) support, the updates, just not the RHEL bill :) Fedora is way too experimental for any system you would want to be stable IMHO. Choosing any free RedHat EL rebuild is a safe, conservative and widely supported choice and easy to install (lots of docs) Cheers! On Wed, 5 Jan 2005, Lane wrote: Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind around. It's a completely different landscape! And why is it that /etc/modules.conf in one release is /etc/modprobe.conf in another? Is this beast not based on standards? But I digress. I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did. At least with unix I was able to get a dial tone! Not so much with this flavor of linux. Each time I run modprobe wcfxs I get the following errors in /var/log/messages: Jan 5 17:57:59 asterisk wait_for_sysfs[2782]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2784]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap2' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2786]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap3' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] Jan 5 17:57:59 asterisk wait_for_sysfs[2788]: either wait_for_sysfs (udev 039)needs an update to handle the device '/class/zaptel/zap4' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to [EMAIL PROTECTED] I'm not so interested in notifying these guys at lists.sourceforge.net, since I'm only interested in running asterisk. Once I commit to actually using linux I might participate in their forum, but not yet :) So ... the question: What flavor of linux does asterisk actually run on Out the box? I'm not scared to compile asterisk, but I'm not at all interested in recompiling a linux kernel. Of course if that is the only way, then I guess I'll just bite another bullet. Hell! I want this PBX to work so bad that I can almost taste it! Please advise. lane ___ 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] TDM04B vs Dell
I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might 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
Re: [Asterisk-Users] ASTCC Compiling Problem
You need some more perl modules. DBI and DBD-mysql I believe. Darren Wiebe [EMAIL PROTECTED] Rafael J. Risco G.V. wrote: I have this error compiling ASTCC: [EMAIL PROTECTED] astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate DBI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 46. BEGIN failed--compilation aborted at ./astcc.agi line 46. make: *** [install] Error 2 [EMAIL PROTECTED] astcc]# I have also installed asterisk-perl from http://asterisk.gnuinter.net/ but i get same error ... any idea?? Rafael ___ 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] Polycom IP500 - problems with multiplesimultaneous calls
I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first is busy, you have to configure that in Asterisk and the phone so that it moves to the next Dial command on a busy from the phone. However, to get a busy from the phone, you have to configure the phone not to have the second ringing instance for the single registration. Confused? I was... So... Your phone display looks some6thing like this... LINE 1 LINE 2 BLAH In order to make line 1 toll into line 2, you need to make the phone SIP back that it is busy when Line 1 is engaged. Default out of the box is to show the incoming call on line below the current call on line one. This will be confitgured in the mac-phone.cfg file for your phone. Probably in this section... divert divert.1.contact= divert.1.autoOnSpecificCaller=1 divert.2.contact= divert.2.autoOnSpecificCaller=1 divert.3.contact= divert.3.autoOnSpecificCaller=1 divert.4.contact= divert.4.autoOnSpecificCaller=1 divert.5.contact= divert.5.autoOnSpecificCaller=1 divert.6.contact= divert.6.autoOnSpecificCaller=1 fwd divert.fwd.1.enabled=0 divert.fwd.2.enabled=0 divert.fwd.3.enabled=0 divert.fwd.4.enabled=0 divert.fwd.5.enabled=0 divert.fwd.6.enabled=0/ busy divert.busy.1.enabled=0 divert.busy.1.contact= divert.busy.2.enabled=0 divert.busy.2.contact= divert.busy.3.enabled=0 divert.busy.3.contact= divert.busy.4.enabled=0 divert.busy.4.contact= divert.busy.5.enabled=0 divert.busy.5.contact= divert.busy.6.enabled=0 divert.busy.6.contact=/ noanswer divert.noanswer.1.enabled=0 divert.noanswer.1.timeout=60 divert.noanswer.1.contact= divert.noanswer.2.enabled=0 divert.noanswer.2.timeout=60 divert.noanswer.2.contact= divert.noanswer.3.enabled=0 divert.noanswer.3.timeout=60 divert.noanswer.3.contact= divert.noanswer.4.enabled=1 divert.noanswer.4.timeout=60 divert.noanswer.4.contact= divert.noanswer.5.enabled=0 divert.noanswer.5.timeout=60 divert.noanswer.5.contact= divert.noanswer.6.enabled=0 divert.noanswer.6.timeout=60 divert.noanswer.6.contact=/ dnd divert.dnd.1.enabled=0 divert.dnd.1.contact= divert.dnd.2.enabled=0 divert.dnd.2.contact= divert.dnd.3.enabled=0 divert.dnd.3.contact= divert.dnd.4.enabled=0 divert.dnd.4.contact= divert.dnd.5.enabled=0 divert.dnd.5.contact= divert.dnd.6.enabled=0 divert.dnd.6.contact=/ /divert Note the No Answer and Busy Fwd parameters. These will be yor starting point and should be docuented in the Polycom Admin PDF. The * side should just be an extension definition that moves from one Line to the Next if busy. The bad news is it seems like some of the features for these phones just don't work so I cannot guarantee that what I sent will work. I finally gave up on this for lack of tiem and just registered a single Line and let the phone ring in a second line to the extension. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, January 05, 2005 5:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP500 - problems with multiplesimultaneous calls Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the on-hold feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my dialplan go to the second appearance/registration. Instead, the second call rings on the first line, and allows you to put the first call on hold, and take the second call. To do so, though, you have to press the little down arrow and then press Answer. When the third call comes in, it will ring the 2nd line. I find this to be non-intuitive, but I can get used to it. My receptionists, however, are finding it REALLY painful. I'd just like to make the first call go to line appearance 1, the second simultaneous call to go to line appearance 2, etc. Maybe somebody figured out a neat dialplan thing to get this done. My config that doesn't do what I want looks like this: ; The first line appearance is registered to 18, the second to 1802, and the third to 1803 exten = 18,1,Dial(SIP/18,20) exten = 18,2,Voicemail(u18) exten = 18,102,Goto(1802,1) exten = 1802,1,Dial(SIP/1802,20) exten = 1802,2,Voicemail(b18) exten = 1802,102,Goto(1803,1) exten = 1803,1,Dial(SIP/1803,20) exten = 1803,2,Voicemail(b18) exten = 1803,102,Voicemail(b18) exten = 1803,103,Hangup I guess the phone just doesn't register as busy when there is
[Asterisk-Users] modprobe: Can't locate module wctdm
After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we use it daily for calls in our office. I am stuck.. Any help? I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this time before Zaptel, is that bad? Thanks, Todd ___ 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] CallerID in Australia Analogue PSTN Phone
On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits
into the following:
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
Sure was. It was me calling myself from my mobile (cell) phone, and
that definitely has CLID enabled. In AU CLID is enabled by default.
Only for mobiles, and that's incoming. It's not enabled for landlines by
default (at least for landlines that were around since before callerid
was introduced ~5 years ago). For testing that sort of thing picking up
a $30 clid box might be worth it.
Do you know if the Digium X101P has problems with reading CLID on the
line? There is a wiki that says that in AU the DEFAULT_CIDRINGS needs
to be =2 rather than the default =1 and I have set that; perhaps I
should reverse that and try again.
pgpzQZ0fTwKy1.pgp
Description: PGP signature
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Re: [Asterisk-Users] Out the box solutions?
On Wed, 2005-01-05 at 18:14 -0600, Lane wrote: Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind around. It's a completely different landscape! And why is it that /etc/modules.conf in one release is /etc/modprobe.conf in another? Is this beast not based on standards? But I digress. I chose FC3 (Fe Personally, I've never had a problem with Debian. Use the testing version (as opposed to unstable or stable). I suspect this should work quite painlessly. PS, In my experience, RH is quite 'loose' with the 'standards' while debian tends to be quite strict. Also, regardless of which distro you choose, it will be quite different to FreeBSD, I walked into a job with a bunch of FreeBSD servers having only experience on Linux, and, while it is different, unusual, and seemingly stupid at times how things are done, eventually you will see the good sides as well. off topic rambling There is something about make world that I just really liked However, I never did like the lack of information in /proc on FreeBSD, nor the lack of lots of 'standard (on linux)' tools that are so helpful. So, overall, there are some things better done in linux, and some better in FreeBSD, but since we can't live on both sides of the fence, you gotta take the good with the bad... /off topic rambline Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID in Australia Analogue PSTN Phone
On Thu, Jan 06, 2005 at 12:14:12PM +1100, Julien Goodwin wrote: On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits into the following: Sure was. It was me calling myself from my mobile (cell) phone, and that definitely has CLID enabled. In AU CLID is enabled by default. Only for mobiles, and that's incoming. It's not enabled for landlines by default (at least for landlines that were around since before callerid was introduced ~5 years ago). For testing that sort of thing picking up a $30 clid box might be worth it. The predominant carrier default for landlines is CLID sending enabled and CLID reception disabled. You can change the sending default on your lines without charge, but reception requires monthly payment. Other carriers may differ. -- Christopher Vance ___ 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] CVS Compile problem on Solaris
Hello all, After reading through the Wiki and archives, I decided to take a stab at installing * on Solaris 9 SPARC. I checked it out via CVS last night as well as about an hour ago, and have run into a compile problem that I can't quite figure out. After running into some unlisted dependencies, such as popt, things are almost compiling. Right now the make bombs when trying to find setenv (which is prototyped in the include/solaris-compat/compat.h) and unsetenv (which is not prototyped in the solaris-compat) when compiling/linking smsq. I cannot find a manual entry or match on google for setenv on solaris aside from the shell command. Is there another prerequisite package that I'm missing, or a linker path that I should add? A transcript of the problem is below. Thanks for any pointers! --Max ... make[1]: Entering directory `/ext2/asterisk/utils' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6-Wcast-align -DSOLARIS -DASTERISK_VERSION= \CVS-HEAD-01/05/05-16:40:01\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR= \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH= \/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules \ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNO_AST_MM -c -o smsq.o smsq.c smsq.c: In function `txqcheck': smsq.c:103: warning: int format, pid_t arg (arg 4) smsq.c: In function `rxqcheck': smsq.c:173: warning: int format, pid_t arg (arg 4) smsq.c:197: warning: implicit declaration of function `unsetenv' smsq.c:215: warning: implicit declaration of function `setenv' smsq.c: In function `main': smsq.c:650: warning: int format, pid_t arg (arg 4) smsq.c:653: warning: int format, pid_t arg (arg 7) gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6-Wcast-align -DSOLARIS -DASTERISK_VERSION= \CVS-HEAD-01/05/05-16:40:01\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR= \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH= \/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules \ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNO_AST_MM -o smsq smsq.o -lpopt Undefined first referenced symbol in file unsetenvsmsq.o setenv smsq.o ld: fatal: Symbol referencing errors. No output written to smsq collect2: ld returned 1 exit status make[1]: *** [smsq] Error 1 make[1]: Leaving directory `/ext2/asterisk/utils' make: *** [subdirs] Error 1 ___ 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] TDM04B vs Dell
TDM400's use the wcfxs module to drive both FXO and FXS ports on them. I have an IBM xSeries 305 (1U P4 2.4ghz 1gb of RAM) server and I just picked up a TDM04B today, and I am getting the exact same problem. When I make calls to/from the TDM04B card I get this really really staticky sound. Calls show up however. Any resolutions to this? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, January 05, 2005 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM04B vs Dell I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might 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 ___ 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] TDM04B vs Dell
At 06:53 PM 1/5/2005 -0600, you wrote: I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. I agree. Perhaps I have too much faith in Digium support. Does anyone else disagree with Digium's assessement? The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? Actually, I tried modprobe wctdm which was supposed to load the correct TDM driver and this resulted in the same behavior described above (lights on, system locks.) In an attempt to figure out why the system locked up I subsequently issued a modprobe wcfxs to confirm that was causing the problem. I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might look. Thanks for the advice. However, modprobe zaptel didn't do anything (that I could tell) and modprobe wcfxo returned the error. And, greping for Fedora in src/zaptel didn't turn up any matches. Michael Swan Neon Software, Inc. ___ 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] modprobe: Can't locate module wctdm
On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote: After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we use it daily for calls in our office. I am stuck.. Any help? I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this time before Zaptel, is that bad? I had this happen myself just recently. Moreover, I've never been able to get modprobe wctdm to work. I'm running v1.0.3 on FC1. I end up doing modprobe zaptel modprobe wcfxs ztcfg -vv which makes it all work. However, on Monday I restarted my server and modprobe could not find zaptel at all. I ended up doing a cvs update of zaptel and it finally started ok. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] Sending DTMF to PSTN problem with SIP
CClarke wrote: Dear All ~ I have * setup running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the SIP2000 to use inband dtmfmode (as opposed to auto), and likewise in sip.conf, but no success. btw, I've also set relaxdtmf=yes in zapata.conf since inbound calls sometimes seem to have trouble dialling extensions. A soft IAX phone (e.g. DIAXPhone) works ok, so I suspect my SIP2000/sip.conf setup, but can't see what I'm doing wrong. You can only use inband DTMF when using the ulaw or alaw codec. If you use any other codec you need to use Out of Band DTMF, i.e. INFO or RFC2833. This is not an Asterisk issue. Codecs discort DTMF. Both Asterisk and the phone should be set to the same DTMF mode. ___ 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] Streaming Audio - Music On Hold Feature
On Wed, 5 Jan 2005 16:50:12 -0700 (MST), Dan Adams [EMAIL PROTECTED] wrote: I was wondering, does anyone know if it is possible to have a stream of audio coming from a Microsoft compressed audio stream fed to the caller if they are placed on hold and if so how might this be done? I have not used any M$ products, but it works with shoutcast like this: default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://host.com:8000/ basically, create an empty directory to point it to first, then the url to the stream. If the microsoft stream can be played via url in winamp in MP3 format, then it should work about the same. ___ 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] modprobe: Can't locate module wctdm
I get the same error with modprobe wcfxs. It's weird, yesterday I installed CVS Head and the latest Zaptel and did not have these problems.. I tried updated Zaptel via CVS then ran make clean; make install. When I tried modprobe wctdm it still flaked and when I tried to start asterisk it totally just blew up.. Since I have a little time at the moment and I am trying to learn from this, I started over and formatted :) Now I am wondering which version of Asterisk and which version of Zaptel I should get this time... I want a stable release of Asteisk, not the latest CVS but I am not sure if I need a matching version of Zaptel or if I can and should get the latest version of Zaptel for this newer analog card I have. Does anyone know: 1- If the version of Zaptel and the version of Asterisk MUST be the same? 2- If I need the latest version of Zaptel to run the TDM400 card? Thanks! - Original Message - From: Michael Graves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 05, 2005 7:43 PM Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote: After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we use it daily for calls in our office. I am stuck.. Any help? I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this time before Zaptel, is that bad? I had this happen myself just recently. Moreover, I've never been able to get modprobe wctdm to work. I'm running v1.0.3 on FC1. I end up doing modprobe zaptel modprobe wcfxs ztcfg -vv which makes it all work. However, on Monday I restarted my server and modprobe could not find zaptel at all. I ended up doing a cvs update of zaptel and it finally started ok. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] modprobe: Can't locate module wctdm
Also, I wonder if there is some sort of issue with the fact that I compiled and installed Asterisk before Zaptel? ?? - Original Message - From: Michael Graves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 05, 2005 7:43 PM Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote: After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we use it daily for calls in our office. I am stuck.. Any help? I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this time before Zaptel, is that bad? I had this happen myself just recently. Moreover, I've never been able to get modprobe wctdm to work. I'm running v1.0.3 on FC1. I end up doing modprobe zaptel modprobe wcfxs ztcfg -vv which makes it all work. However, on Monday I restarted my server and modprobe could not find zaptel at all. I ended up doing a cvs update of zaptel and it finally started ok. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] TDM04B vs Dell
I dug around and found my newest UpdateXpress cd from IBM and ran it on this box and updated the BIOS and my problem went away. *shrugs* -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Wednesday, January 05, 2005 7:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM04B vs Dell At 06:53 PM 1/5/2005 -0600, you wrote: I've struggled for several days trying to get a Digium TDM04B 4-port wxfco card working on a Dell 1U PowerEdge 750 machine running Fedora Core 1. I finally got a call back from Digium who indicated that there is a fundamental conflict between the card and the PowerEdge having to do with PCI interrupts. Asterisk version is stable v1-0 12/29/04. That sounds a little hard to believe. I agree. Perhaps I have too much faith in Digium support. Does anyone else disagree with Digium's assessement? The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? Actually, I tried modprobe wctdm which was supposed to load the correct TDM driver and this resulted in the same behavior described above (lights on, system locks.) In an attempt to figure out why the system locked up I subsequently issued a modprobe wcfxs to confirm that was causing the problem. I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. I don't use Fedora, but it seems those that do have had problems loading the drivers. Try the modprobe wcfxo then zaptel, then check your /proc/interrupts. If that doesn't work, try modprobe zaptel only. I think someone mentioned a readme in the src/zaptel directory for Fedora as well. Might look. Thanks for the advice. However, modprobe zaptel didn't do anything (that I could tell) and modprobe wcfxo returned the error. And, greping for Fedora in src/zaptel didn't turn up any matches. Michael Swan Neon Software, Inc. ___ 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] Sending DTMF to PSTN problem with SIP
Thanks for the suggestion! I have tuned the SIPURA SPA 2000 gain as suggested and also re-adjusted X100P gain and echo cancel. That helped a lot with audio quality. But still cannot send DTMF tones out over PSTN. As well I discovered DTMF transmission from IAX soft phone out to PSTN is not at all reliable either, so it seems to be on the PSTN (X100P) side somewhere. I cannot find any settings to tweak DTMF transmission other than relaxdtmf, and can't find any log messages which indicate anything is going wrong. A plain analog phone plugged into the PSTN sends DTMF which is recognized ok. Any ideas appreciated... You can configure the gain to be lower on the SPA2000 via the web interface - Ido not remember the exact location, but you will find it under advanced settings. --- CClarke [EMAIL PROTECTED] wrote: Dear All ~ I have * setup running ok (with two Wildcard X100P's to PSTN). I also have two analog phones connected into same through a SIPURA 2000. These work fine, except that when I call out through PSTN try to send DTMF tones to (say) a remote PBX to dial an extension, the gain seems to go wild (high), and the DTMF tones are not recognized at the other end. I tried setting the SIP2000 to use inband dtmfmode (as opposed to auto), and likewise in sip.conf, but no success. btw, I've also set relaxdtmf=yes in zapata.conf since inbound calls sometimes seem to have trouble dialling extensions. A soft IAX phone (e.g. DIAXPhone) works ok, so I suspect my SIP2000/sip.conf setup, but can't see what I'm doing wrong. Christina. ___ 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] TDM04B vs Dell
On Wed, 2005-01-05 at 18:53 -0600, Rich Adamson wrote: The symptoms of the problem were as follows: 1. issue modprobe zaptel which immediately returns with no feedback Right, there isn't any output from loading this module. 2. issue modprobe wcfxo which returns init_module: No such device Hint: isnmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters Correct, you don't have any cards that need this driver, so it error'ed and refused to load - good. 3. issue modprobe wcfxs which immediately returns with no feedback, however the four lights on the card go on and then the machine locks up completely, requiring a power cycle to get it running again. After the power cycle, if I look in /var/log/messages If you have a tdm04b, that says you have four fxo ports. Why are you trying to load wcfxs? No, ALL modules on the TDM card use the wcfxs module, or in CVS the wctdm module (renamed because of the above mis-understanding). I see a long cycle of the following messages before reboot: kernel: Dazed and confused, but trying to continue kernel: Do you have a strnage power saving mode enabled? kernel: Uhhuh, NMI received for unknown reason 20 on CPU 0 So, this is the problem 4. if I cat /proc/interrupts, I don't see any entry for a wcfxo module. Again, you shouldn't because the module never loaded because you don't have a X101P card installed. In any case, I did follow the setup instructions on the Digium site (make install in /usr/src/zaptel, edit /etc/zaptel.conf, edit /etc/asterisk/zapata.conf, etc.) and we currently have a X100P wcfxo card in another machine running well so we've already had experience getting a card working. If anyone has insight into what might be wrong, please do let me know. Ultimately, if I trust the Digium support information, then this card will never work, so I'd be grateful to hear about any other PCI card that provides four or so wcfxo interfaces that might work with the PowerEdge. Well, IMHO, there are two option, blame digium for producing a card that doesn't meet the PCI spec, or blame Dell for producing a motherboard that doesn't meet the PCI spec (or whatever is the spec that is causing this problem). If it is Dell, then harass them to either give you a different server, or just return it and purchase a different server from someone else, or harass dell to fix the problem (if it is BIOS software then it should be easier)... This same conversation applies to the recent thread about some HP/Compaq server models. We need to determine who is 'at fault' not so we can point the finger etc, but so we know either to avoid Dell model xxx or Compaq yyy, or, that digium have got it wrong, and maybe they can provide some sort of new hardware version which fixes it, or a software workaround, or something. Just my 0.02c worth. PS, either way, all of this should be resolved as much as possible with digium directly, not on this list. Also, regardless of how good/bad your experience in dealing with digium support, consider that it isn't going to help your case by bad mouthing them on the list, nor is it really pertinent to your problem. So just skip it. Currently, you might say they have an excuse due to their recent holidays, plus probably some sort of backlog due to holidays. However, Digium should probably take note of these comments and try to do something about this. Also, like any other company, if you are not satisfied with the level of customer service, then write a letter of complaint to the *relevant* person. Not this list. There, that's 0.04c worth now. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Cisco 7920 and Asterisk - Anyone got this working?
I'm not entirely sure this phone supports sip. Have you tried building the asterisk extra's and configuring it with skinny? Erik On Thu, 6 Jan 2005 00:37:32 -, Paul A Brown [EMAIL PROTECTED] wrote: I am having all sorts of probs. It just won't connect. Anyone got any example configs I could look at? Thanks Paul ___ 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] Aaargh Gentoo updated some packages now * won't start
On Wed, 5 Jan 2005 Remco BarendeB wrote: After emerging some updates this morning asterisk 1.0.3 fails to start I get the following errors: CUT Check your file permissions. * recently got pretty picky about them lately -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ 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] modprobe: Can't locate module wctdm
Yes, I believe that this is a problem. Everything I've read says you compile and install zaptel first...then asterisk. On Monday I rebooted my server again, the just did a CVS update of zaptel. That was all the was required. Michael On Wed, 5 Jan 2005 20:10:27 -0600, Me wrote: Also, I wonder if there is some sort of issue with the fact that I compiled and installed Asterisk before Zaptel? ?? - Original Message - From: Michael Graves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 05, 2005 7:43 PM Subject: Re: [Asterisk-Users] modprobe: Can't locate module wctdm On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote: After getting zaptel from the CVS server, compiling and installing it I type: modprobe zaptel and all is well. Then I type: modprobe wctdm and I get this: modprobe: Can't locate module wctdm Any idea why? I did this yesterday but with the CVS head of Asterisk and I got by this part without a problem. I reinstalled it all today because I wanted the stable release on our server since we use it daily for calls in our office. I am stuck.. Any help? I am running Fedora 1 FYI.. Also, I compiled and installed Asterisk this time before Zaptel, is that bad? I had this happen myself just recently. Moreover, I've never been able to get modprobe wctdm to work. I'm running v1.0.3 on FC1. I end up doing modprobe zaptel modprobe wcfxs ztcfg -vv which makes it all work. However, on Monday I restarted my server and modprobe could not find zaptel at all. I ended up doing a cvs update of zaptel and it finally started ok. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
