Re: [Asterisk-Users] Intel Modem vs Digium Cards
On Mon, 11 Oct 2004 00:10:26 +0100, David J Carter [EMAIL PROTECTED] wrote: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave £55 is more like US$100 :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Error starting
Simon Brown wrote: I have just downloaded V1.0 from CVS and when I try to start Asterisk (after compiling and installing) I get this error: Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get parameters Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to register channel '2' Oct 11 15:51:29 WARNING[1076241024]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Oct 11 15:51:29 WARNING[1076241024]: loader.c:429 load_modules: Loading module chan_zap.so failed! Does anyone have any ideas what is wrong? Did you upgrade to V1.0, or is this a brand-new installation? New install: Do you have any digium cards on the machine? If not then that's probably why it's throwing up the 'Loading module chan_zap.so failed' error. If you're going to run the * box without any of the digium cards then you need to add noload = chan_zap.so in /etc/asterisk/modules.conf New install, have digium card: Perhaps your box isn't recognizing the card due to an IRQ issue? run the dmesg command and make sure there's something in the output that shows your box recognizes the existence of the card. Perhaps posting a bit more information about your setup might help. Cheers Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Error starting
I have been running Asterisk happily for many months and I was trying to upgrade from CVS-HEAD-08/13/04-10 Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of el Flynn Sent: Monday, 11 October 2004 16:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Error starting Simon Brown wrote: I have just downloaded V1.0 from CVS and when I try to start Asterisk (after compiling and installing) I get this error: Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get parameters Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to register channel '2' Oct 11 15:51:29 WARNING[1076241024]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Oct 11 15:51:29 WARNING[1076241024]: loader.c:429 load_modules: Loading module chan_zap.so failed! Does anyone have any ideas what is wrong? Did you upgrade to V1.0, or is this a brand-new installation? New install: Do you have any digium cards on the machine? If not then that's probably why it's throwing up the 'Loading module chan_zap.so failed' error. If you're going to run the * box without any of the digium cards then you need to add noload = chan_zap.so in /etc/asterisk/modules.conf New install, have digium card: Perhaps your box isn't recognizing the card due to an IRQ issue? run the dmesg command and make sure there's something in the output that shows your box recognizes the existence of the card. Perhaps posting a bit more information about your setup might help. Cheers Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: Asterisk-Users Digest, Vol 3, Issue 121
dean collins [EMAIL PROTECTED] writes: Lol, you're kidding right, go and look at what it costs to buy an alternative ip-pabx in comparison, and sorry but no corporate budget here, this is just a system for my home $100 on an old P3-700, and about the same on a card, and 2 $55 grandstream handsets along with some free sip softphone software. Hardly a fortune. If I were looking for a home system I would agree with you (and I also agree with the sentiment expressed by numerous posters that Digium deserves all the support it gets), and I am glad that you have no trouble spending $310 on your home system. I am doing technical support for a non-profit K12 school serving aid workers, refugees and others of relatively low income. I cannot spend a lot of money on investigating new technology; I need to do my investigation and testing on a shoestring budget. I realize that a commercial ip-pabx (or really any commercial pbx) would cost much more; (a) that is why I am looking at asterisk; (b) I would not need to test anything; I'd buy it, and if something doesn't work it's the vendor's problem, not mine. On the other hand I think we are very fortunate that asterisk exists and cant help but get excited about where they will grow to. True. And for that reason I wish Digium all the best, and will buy my production hardware from them. When I have tested the system and become sufficently familiar, using hardware I can afford. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Error starting
From what I have seen so far on this list if you are running a version of CVS-Head prior to release of Asterisk 1.0 then you should keep it and not try to change or upgrade it. It would appear that there are a lot of recent changes that may break if you try to upgrade to current CVS-Head, and conversely that 1.0 is missing a lot of functionality that was present in August / September CVS-Head. Craig - Original Message - From: Simon Brown [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 2:16 PM Subject: RE: [Asterisk-Users] Error starting I have been running Asterisk happily for many months and I was trying to upgrade from CVS-HEAD-08/13/04-10 Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of el Flynn Sent: Monday, 11 October 2004 16:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Error starting Simon Brown wrote: I have just downloaded V1.0 from CVS and when I try to start Asterisk (after compiling and installing) I get this error: Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get parameters Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to register channel '2' Oct 11 15:51:29 WARNING[1076241024]: loader.c:334 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Oct 11 15:51:29 WARNING[1076241024]: loader.c:429 load_modules: Loading module chan_zap.so failed! Does anyone have any ideas what is wrong? Did you upgrade to V1.0, or is this a brand-new installation? New install: Do you have any digium cards on the machine? If not then that's probably why it's throwing up the 'Loading module chan_zap.so failed' error. If you're going to run the * box without any of the digium cards then you need to add noload = chan_zap.so in /etc/asterisk/modules.conf New install, have digium card: Perhaps your box isn't recognizing the card due to an IRQ issue? run the dmesg command and make sure there's something in the output that shows your box recognizes the existence of the card. Perhaps posting a bit more information about your setup might help. Cheers Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where did USE_SIP_MYSQL_FRIENDS go?
There is no USE_MYSQL_FRIENDS and USE_SIP_MYSQL_FRIENDS in .../asterisk/channels/Makefile any more. But, on voip-info wiki it still says that it should be configured like this. Anyone knows how should I tell Asterisk to use mysql database for SIP and IAX friends? Thanks Tomica Crnek ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Intel Modem vs Digium Cards
Hello, this is not really much of an issue any more in Europe, the old state-owned monopoly phone companies have had to loosen up in the face of private competition and de-regulation (or rather, fairly liberal re-regulation). I something I hook up causes an actual technical malfunction in the switch, the telco will turn my line on and might charge me for any actual damage, but otherwise I am free to use whatever hardware I want. But your descriptions of the situation in Japan (i.e. a week ago or so on the way NTT sells/leases pbxs) is very much the way it was here as little as 8-10 years ago. And it is not that long ago that things were that way in the US, but we tend to forget that rather quickly (if we ever were aware of it). Ten years ago if you had a PBX connected to PSTN you had to have a support contract with a licensed vendor ... four years ago the telco sold us an Alcatel PBX and couldn't care less that we didn't want to pay for a support contract. But of course, since PBXs were not designed to be customer-maintained, there is virtually no documentation available. But things are changing. Regards, Wolf Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes: From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards To: Brian West [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII On Sun, 10 Oct 2004 13:28:46 -0500, Brian West [EMAIL PROTECTED] wrote: (Benjamin) Having said that, you have a good case in favour of the Intel modems if you are in a country where the X100P doesn't have type approval but you can find an Intel modem (with the right chipset) that does. In such a case, using the Intel modem might be the only legal way to connect your Asterisk box to an analog PSTN line. (Brian West) Not really the X101P is really just a modem that already has the approvals. They stick a heatsink of the md3200 chip and call it an x101p. (Benjamin) You are mistaken. The approval is given for a certain production run of a certain design, not for the chipset nor for any similarly designed modem. It's got to be the exact same make. Here in Japan for example, there used to be a Taiwan made modem based on the Intel/Ambient chipset which has type approval while Digium's X100P does not. The manufacturer of this modem released an updated model which is not imported to Japan anymore and that updated model does not have type approval even though it is just a slightly different version of its approved predecessor. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream phone price
Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] h.323 debian sarge problem - Could not open sound channel
Jeremy McNamara wrote: Mészáros Mihály wrote: Please if you can please help me to solve this problem. Help yourself and READ THE README. Hello Jeremy! I read it already! ;-) thx! But i didn't find a word about that chan-h323 what decoder encoder use. It use the libopenh323 or other in built encoder ? I have problem (as you can see in my trace) in opening libopenh323 encoder. My question was can i override function OpenAudioChannel original is in libopenh323 - h323ep.cxx function in chan-h323 MyH323EndPoint can i ignore opening sound device ? Or chan-h323 use it somehow? Regards, Misi Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Reload Asterisk from php or perl script
Hi,
I am looking for a basic script that can reload asterisk from
php or perl via a web browser.
I have tried exec( asterisk -rx reload ) and shell_exec( same cmd )
with php but there seems to be a permission issue with asterisk that
stops these working. I was just wondering if anyone has a way around
this with perl or php.
besides I prefer to use the manager, cause is more secure,
easy, etc, another way to reload from php is to call the
script with a wrapper in perl, like:
test.php is the script that does fancy things and contains
something like asterisk -rx reload somewhere,
and /or writes * config files, blah blah...
the test perl script would be something like:
#** cut here
#!/usr/bin/perl
# Perl wrapper to execute a PHP script setuid
# Requires PHP CLI
use File::Basename;
# Make
UID = EUID (so that PHP can run system()s and execs() setuid)
$ = $;
# Set
this to the path, so that we can't get poisoned
$ENV{'PATH'} = /var/lib/asterisk/scripts;
$ENV{'BASH_ENV'} = /var/lib/asterisk/scripts;
# Open
the PHP script
$data = basename($0);
if
($data =~ /^([EMAIL PROTECTED])$/) {
$data = $1; # $data now untainted
} else {
die Bad data in $data;# log this somewhere
}
system($data..php);
#** cut here
and call /var/lib/asterisk/scripts/test
btw, the manager is better :)
Matteo.
--
Matteo Brancaleoni
System Administrator
[EMAIL PROTECTED]
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:[EMAIL PROTECTED]
Tel. +39 0270633354
Fax. +39 0245487890
IAXTEL: 17005662458
http://www.espia.it
___
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RE: [Asterisk-Users] Grandstream phone price
$1.64 to the £1 I think this morning so $35 stands. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Grandstream phone price
Forget the last post, the brain is totally screwed. Must get more sleep. Thanks all for pointing the errors of my conversion, so used to working the other way. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Intel Modem vs Digium Cards
On Sun, 2004-10-10 at 22:32 -0500, Steven Critchfield wrote: On Mon, 2004-10-11 at 00:10 +0100, David J Carter wrote: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Your conversion above is going the wrong way. a British pound is worth more than a US Dollar. In fact, 55 British pounds is nearly $100USD now. Look at http://www.xe.com/ucc/convert.cgi No Steve, He works for NASA. :) -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Registering to H323 Gatekeeper as client
oi geli wrote: Can I use the Asterisk to register to a H323 Gatekeeper as client? I have the GK IP address and the user id. I am using chan-h323 (from CVS). Please share the h323.conf if you have it working. I did not see any GK user id field in the h233.conf. Thanks We are using * as SIP, H323 and MGCP translator. You have sample h323.conf as attachment. [general] port = 1720 bindaddr = 0.0.0.0 ;allow=g729 allow=all ; turns on all installed codecs ;dtmfmode=rfc2833 dtmfmode=inband gatekeeper = 10.0.0.250 AllowGKRouted = yes context=from-h323 [asteriskgw1] type=h323 prefix=12 e164=110 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Intel Modem vs Digium Cards
On Mon, 11 Oct 2004 09:23:54 +0200, Dave Cotton [EMAIL PROTECTED] wrote: No Steve, He works for NASA. :) hilarious :-) This reminds me of an anecdote I'd like to share ... After WWII, US Army officials set new values for measurement units in defeated Japan. At some point they came to a unit Yen, the character of which can also be translated into circle when taken out of its monetary context. The army officials quickly concluded that the new value for the unit circle should be 360 degrees and so the Yen-Dollar exchange rate was fixed at 360 yen to the dollar. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream phone price
On Mon, 11 Oct 2004 08:35:06 +0100, David J Carter [EMAIL PROTECTED] wrote: $1.64 to the £1 I think this morning so $35 stands. I can only hope you are not working on any billing software ;-) rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream phone price
You multiply to get the dollar price. Careful where you go on holiday, it could be costing more than you think!! -Original Message- From: David J Carter [mailto:[EMAIL PROTECTED] Sent: 11 October 2004 08:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Grandstream phone price $1.64 to the £1 I think this morning so $35 stands. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request for IAX debug session transcript with IAXy
Hi can somebody who has got an IAXy please run a debug on their Asterisk server and send me the session transcript of an attended transfer (assuming the IAXy supports this) ? I am currently creating call flow charts for IAX call scenarios to assist a phone manufacturer to implement IAX and support it on their phones. The charts will also go into Frank Miller's IAX specification document. What I am looking for is the IAX transcript of the following scenario: A is an IAX device with an established PSTN call to C which is going through Asterisk server B ... [A]---IAX---[B]---Zaptel-PSTN---[C] Now A parks the call with C and makes a new call to D ... [A]---IAX---[B]---SIP---[D] Finally, A requests B to transfer the call with C to D after which A hangs up ... [C]---Zaptel-PSTN---[B]---SIP---[D] thanks in advance rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Grandstream phone price
In article [EMAIL PROTECTED], David J Carter [EMAIL PROTECTED] wrote: $1.64 to the £1 I think this morning so $35 stands. Dave So that makes £55 to be 55 x $1.64 = $90.20 It wasn't the rate that was at issue, but that the OP divided instead of multiplying. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: bt communicator`
Hi Robert; First, you have to use the SIP2 channel code (chan_sip2.c) from http://bugs.digium.com/bug_view_page.php?bug_id=759 as this does the proxy-authenticate properly. Get the module, follow the build instructions, and add noload=chan_sip.so to stop the old code loading. It will autoload the new one. You need to know the username that the Yahoo Communicator uses. Ethereal or similar will trace SIP for you. The username I type into communicator has .brz appended by the Communicator for some reason. The password is the one you type into communicator but I had to MD5 it. Comment below password shows how. [general] ;port = 5060; Port to bind to port = 5052 ; change to 5052 as 5060 will not authorise on BTCommunicator ; Note if you want local SIP on 5060, you need to use siproxd or similar to redirect (unless anyone knows otherwise) pedantic=no disallow=all; Disallow all codecs allow=alaw ; Allow codecs in order of preference ; BT uses a-law allow=ulaw allow=gsm ;allow=ilbc defaultexpirey=1200 ; Change for BT as it objects to 3600 - note deliberate spelling error register = [EMAIL PROTECTED]:[EMAIL PROTECTED] ; Need to state externip as the internal address otherwise BT won't work - something to do with NAT ;externip = 195.13x.xx.xx externip = 192.168.10.250 localnet = 192.168.10.0/255.255.255.0 ;. ;. ;. [bt] type=friend nat=yes disallow=all allow=alaw canreinvite=no username=[username].brz authuser=[username].brz fromdomain=btinternet.com fromuser=[username].brz auth=[username]:[EMAIL PROTECTED] ; I didn't have the .brz here and it works? md5secret=6eb36df5f5d94381973b6090b30e0f59 host=btinternet.com ;outboundproxy=sip.btcommunicator.bt.net;not needed ;outboundproxyport=5060 ;not needed ;MD5 ;alambil:/etc/asterisk# echo -n [EMAIL PROTECTED]:btinternet.com:[password] | md5sum ;6eb36fd5f5d94381973b6090b30e0f59 - Once this worked, I didn't change it. There are probably unneeded lines above. Regards Peter -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 09 October 2004 21:40 To: Whisker, Peter Subject: bt communicator` Hi Peter I have been following your post but didn't see the other emails about getting it working until now!! Could you please send me the details for the chan_sip2 method Thanks Robb This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Am I stupid or is my card DOA.?
I had/have exactly the same problem with my X100P / TDM400P dev setup. To fix this error all I did was swap the PCI slots that the cards were in. And this error came back either due to a reboot or because I update to the latest CVS. But again after much messing around with config's and BIOS options I gave up and swapped the cards again (well went from X on 1 - TDM on 2 to X on 0 - TDM 1). Restarted and hey-presto the TDM was recognised. This does have me a little concerned as I'm running out of combinations of PCI slots. HTH Alex -Original Message- From: Martin Marshall [mailto:[EMAIL PROTECTED] Sent: 09 October 2004 22:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Am I stupid or is my card DOA.? Thank you for the information. My kernel was a stock Fedora Core 1 kernel (2.4.22). I have now tried with the latest 2.4.27 kernel with the same results. I will now try to rollback to an earlier release of zaptel code. Regards Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 09 October 2004 16:39 To: Asterisk List Subject: Re: [Asterisk-Users] Am I stupid or is my card DOA.? On Sat, 2004-10-09 at 16:16 +0100, [EMAIL PROTECTED] wrote: I wonder if someone with a little more experience with the zaptel driver could give me a little help with the following driver loading issue. My Setup: One TDM400P PCI Card with 2 x FXS Modules One X100P PCI FXO Card I have downloaded the latest CVS of the zaptel driver and compiled it against my current kernel source. However, each time I load the driver, I get the following error. I have also enclosed a copy of the zaptel.conf, lsmod, lspci and /proc/interrupts. My machine is running Redhat FC1 on Intel(R) Xeon(TM) CPU 2.80GHz's (SMP Kernel). I have also tried the card in an AMD (1xWay) with the same results (no PCI conflicts - that I can see). # modprobe zaptel (Loads OK) # modprobe wcfxo (Loads OK with the normal status information) # modprobe wcfxs /lib/modules/2.4.22-1.2115.nptl.msm/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.22-1.2115.nptl.msm/misc/wcfxs.o: insmod /lib/modules/2.4.22-1.2115.nptl.msm/misc/wcfxs.o failed /lib/modules/2.4.22-1.2115.nptl.msm/misc/wcfxs.o: insmod wcfxs failed # ztcfg ZT_CHANCONFIG failed on channel 2: No such device or address (6) --- lspci -vv --- [EMAIL PROTECTED] zaptel]# lspci -vv 03:01.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device 8085:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at 3000 [size=256] Region 1: Memory at fb24 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 06:01.0 Network controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device a8fd:0001 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 72 Region 0: I/O ports at 5000 [size=256] Region 1: Memory at fc80 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- The answers to your questions are no and probably not. According to lspci you seem to have 2 X100s, and I have had this with the same setup. My question is what version of the kernel are you using? Because I'm experiencing the same type of problems now with a TDM card that worked perfectly before on 2.4 kernels and older drivers. This is either a kernel problem or something in the newer zaptel drivers. I have pulled the TDM and it has worked perfectly in another machine but not when an X100 is involved. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] re: ATA units: anyone have these working with * or SER?
Hello list, please take a look at these units: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596 are they locked? does anyone have one working with asterisk or SER? Are these rebadged units from a different manufacturer? anyone have any experience good or bad with these? thanks, yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 hangup issue
hi if I'm on the phone to somewhere through this SMART IAD SIP/FXS gateway, and I somehow lose contact with the SIP server (for instance the SMART IAD reboots), then the channel will hang until the other part hangs up. is it possible to force a hangup on a channel in which the caller is no longer available? this would be the desired functionalityl. regards roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a reply in order to use sql database for all
sip clients from small offices asterisk box with nat
context.
May I use autocreatepeer in all asterisk sip.conf file
with nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost= Hostname of server
dbuser= Username in MySQL
dbpass= Password for user in MySQL
autocreatepeer=yes
nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall box|--Asterisk
-- +
| SIP peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a écrit :
Hi Harry,
how did you make sip peers on mysql database? is it
working well? where can
I find a documentation so I could migrate my
Asterisk sip config to use
Mysql also.
Regards
Vous manquez despace pour stocker vos mails ?
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database
It says To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS., but there is no MYSQL_FRIENDS in channels/Makefile any more. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Monday, October 11, 2004 11:45 AM To: Glynn Condez Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: SIP peers in MySQL Database Hi, Look at: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers http://www.voip-info.org/wiki-Asterisk+configuration+from+database Is it working well? I don't know because of i'm waiting a reply in order to use sql database for all sip clients from small offices asterisk box with nat context. May I use autocreatepeer in all asterisk sip.conf file with nat=yes in general option ??? [general] dbname= Name of database in your Mysql server dbhost= Hostname of server dbuser= Username in MySQL dbpass= Password for user in MySQL autocreatepeer=yes nat=yes --- -- |Asterisk |-- |nat/firewall box | --- -- | | -- | Internet |-- |nat/firewall box|--Asterisk -- + | SIP peers in |mysql database --- -- |Asterisk |-- |nat/firewall box | --- -- Harry --- Glynn Condez [EMAIL PROTECTED] a crit : Hi Harry, how did you make sip peers on mysql database? is it working well? where can I find a documentation so I could migrate my Asterisk sip config to use Mysql also. Regards Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Crez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arriv ! Dcouvrez toutes les nouveauts pour dialoguer instantanment avec vos amis. A tlcharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] SIP hangup issue
hi if I'm on the phone to somewhere through this SMART IAD SIP/FXS gateway, and I somehow lose contact with the SIP server (for instance the SMART IAD reboots), then the channel will hang until the other part hangs up. is it possible to force a hangup on a channel in which the caller is no longer available? this would be the desired functionalityl. regards roy Use RTP timeout, see sip.conf Freddi ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically loaded)
#
# Copyright (C) 1999, Mark Spencer
#
# Mark Spencer [EMAIL PROTECTED]
#
# Edited By Belgarath Aug 28 2004
# Added bare bones ultrasparc-linux support.
#
# This program is free software, distributed under the
terms of
# the GNU General Public License
#
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a écrit :
It says To enable this, you need to edit the
Makefile in the channels directory of your source
tree and enable MYSQL_FRIENDS., but there is no
MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a écrit :
Hi Harry,
how did you make sip peers on mysql database? is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails ?
Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
Créez votre Yahoo! Mail sur
http://fr.benefits.yahoo.com/
Le nouveau Yahoo! Messenger est arrivé ! Découvrez
toutes les
nouveautés pour dialoguer instantanément avec vos
amis. A
télécharger gratuitement sur
http://fr.messenger.yahoo.com
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Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour
dialoguer instantanément avec vos amis. A télécharger gratuitement sur
http://fr.messenger.yahoo.com
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[Asterisk-Users] Re: Re[2]: cisco ip 7905 legal ..
so, better is to look to another phone, than surcharge cisco ;-) PJ - Original Message - From: AST 386sx Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Monday, October 11, 2004 2:00 AM Subject: Re: Re[2]: cisco ip 7905 legal .. If you are going to buy it new. It should not be a problem at all. Just order the SIP software with the phone. Your order should be somthing like this. CP-7905G - Cisco IP Phone 7905G, Global SW-SMH-UL-7905 - SIP or H.323 license for single 7905 IP phone CON-SNT-CP7905 - 8x5xNBD Svc, Cisco IP Phone 7905 Not the following software(s). SW-CCME-UL-7905 - Cisco CallManager Express License For Single 7905 IP Phone SW-CCM-UL-7905 - CallManager Unit license for single 7905 IP phone Both softwares should be same price. I don't think you can change the software license or type of software even you hav smartnet. --ast386-- - Original Message - From: Danny Zak [EMAIL PROTECTED] To: Kannaiyan Natesan [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: October 10, 2004 3:32 PM Subject: Re[2]: [Asterisk-Users] cisco ip 7905 legal .. Hello Kannaiyan, i need to know the correct procedure; otherwise i will bringing my customers in danger and that is not what i want. i know you can buy the 7905 WITHOUT the callmanager license.. if i load the sip image in it; will that be ok ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: Grandstream price in UK
David J Carter [EMAIL PROTECTED] writes: $1.64 to the £1 I think this morning so $35 stands. But £55 x 1.64 is $90.2, not $35 ... Regards, Wolf ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: ATA units: anyone have these working with * or SER?
Hello list, please take a look at these units: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596 are they locked? does anyone have one working with asterisk or SER? Are these rebadged units from a different manufacturer? anyone have any experience good or bad with these? Here is some additonal information: http://home.businesswire.com/portal/site/google/index.jsp?ndmViewId=news_viewnewsId=20040914005648newsLang=en It appears that the boxes are intended for use with the VOIP2 service. There are fairly detailed manuals here: http://www.voip2.net/callbox.html Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call Transfer Problem
On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with voice menu
Hello all, I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto,contexto_extensiones exten = 2,1,Goto,contexto_operadora The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = 2,2,Background(itranser/trans_operadora) exten = 2,3,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = 1,1,Background(itranser/msg_pasar_ext) My problem, is when I touch the key 1 in my phone, after the msg_bienvenida, asterisk do not pass me to the correct context [contexto_extensiones]. Asterisk do not pass me to any context, asterisk do nothing when I press the 1 key on my phone. Have I missed something in my extensions.conf? or in sip.conf? Thanks Regards from Madrid. Ismael Gil.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Problems with voice menu
I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto(contexto_extensiones,s,1) exten = 2,1,Goto(contexto_operadora,s,1) The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = s,1,Background(itranser/trans_operadora) exten = s,2,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = s,1,Background(itranser/msg_pasar_ext) I've made the corrections to your context's above... Note in particular the Goto command and then using the 's' (start) extension in each extension line, also adjusted the priority numbers. For more info on Goto http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Goto Give that a try and see how you go. Regards, Chris Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetVar() with manager
Hi, I'am intergrating Asterisk with our CRM system. I have tryed this but thats not working: fwrite($socket, Context: local\r\n); fwrite($socket, Setvar: Var\r\n); fwrite($socket, Var: . $channel .\r\n); Regards, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: SIP peers in MySQL Database
From few days ago there is no USE_MYSQL_FRIENDS in channels/Makefile. That is why I
am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark Spencer
[EMAIL PROTECTED] # # Edited By Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux support.
#
# This program is free software, distributed under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a crit :
It says To enable this, you need to edit the Makefile in
the channels
directory of your source tree and enable MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a crit :
Hi Harry,
how did you make sip peers on mysql database? is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails ?
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Re: [Asterisk-Users] Problems with voice menu
Thank you Christopher,
Imade the changes you told me, but, when I try to make an incoming call,
in the Asterisk console, I get
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9'
-- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r")
in new stack
-- Called 501050:[EMAIL PROTECTED]/501050
-- Call accepted by 65.39.205.121 (format ULAW)
-- Format for call is ULAW
-- Accepting AUTHENTICATED call from 65.39.205.121, requested format
= 4, actual format = 4
-- Executing Goto("IAX2/[EMAIL PROTECTED]:4569/14", "incoming|s|1")
in new stack
-- Goto (incoming,s,1)
-- Executing Wait("IAX2/[EMAIL PROTECTED]:4569/14", "1") in new stack
-- Executing Answer("IAX2/[EMAIL PROTECTED]:4569/14", "") in new stack
-- Executing DigitTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "10")
in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "20")
in new stack
-- Set Response Timeout to 20
-- Executing BackGround("IAX2/[EMAIL PROTECTED]:4569/14", "itranser/msg_bienvenida")
in new stack
-- Playing 'itranser/msg_bienvenida' (language 'en')
-- IAX2/65.39.205.121:4569/13 answered SIP/aurelio-92fe
-- Channel 'IAX2/[EMAIL PROTECTED]:4569/14'
unable to transfer
-- Hungup 'IAX2/65.39.205.121:4569/13'
Why I get an "Unable to transfer" error on this channel?
How could I solve this problem?
Any clue will be wellcome
Thanks a lot.
Ismael Gil.
Christopher Lee wrote:
I having a lot of troubles to configure a simple voice menu.In extensions.conf I have the following.[incoming]exten = s,1,Wait(1)exten = s,2,Answerexten = s,3,DigitTimeout,10exten = s,4,ResponseTimeout,20exten = s,5,Background(itranser/msg_bienvenida)exten = 1,1,Goto(contexto_extensiones,s,1)exten = 2,1,Goto(contexto_operadora,s,1)The context refered by the menu. (each context play me a diferent message only )[contexto_operadora]exten = s,1,Background(itranser/trans_operadora)exten = s,2,Dial(SIP/ismael,s,1)[contexto_extensiones]exten = s,1,Background(itranser/msg_pasar_ext)
I've made the corrections to your context's above... Note in particularthe Goto command and then using the 's' (start) extension in eachextension line, also adjusted the priority numbers. For more info on Gotohttp://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoGive that a try and see how you go.Regards,Chris Lee___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Agent monitoring using fop
Is there anyway of monitoring an agent's status using the flash operator panel ? I can monitor a queue easily but seem to hit a brick wall with the agents. Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
it's in there in -r v1-0, but replaced by some realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark Spencer
[EMAIL PROTECTED] # # Edited By Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux support.
#
# This program is free software, distributed under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a écrit :
It says To enable this, you need to edit the Makefile in
the channels
directory of your source tree and enable MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a écrit :
Hi Harry,
how did you make sip peers on mysql database? is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails ?
Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
Créez votre Yahoo! Mail sur
http://fr.benefits.yahoo.com/
Le nouveau Yahoo! Messenger est arrivé ! Découvrez
toutes les
nouveautés pour dialoguer instantanément avec vos
amis. A
télécharger gratuitement sur
http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] Re: SIP peers in MySQL Database
At 13:53 11/10/2004, Roy Sigurd Karlsbakk wrote: it's in there in -r v1-0, but replaced by some realtime stuff in development CVS I haven't found out more about that, though.. Old mysqlfriends is now remove from asterisk. Now you have to use res_config_odbc for setup sip/iax friends. you can read wiki and this file README.extconfig in docs for get more information how to setup it. You will find also example in extconfig.conf.sample Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: ATA units: anyone have these working
please take a look at these units: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596 The price of $30 after rebate certainly looks interesting. are they locked? If the firmware agrees with the manual at http://www.voip2.net/Operator_Manual.pdf , it's not even possible to change the UI password. Also, since this device is intended to go on the WAN side of your router (or to act as a router), some user config is needed, so I would think that it would not be locked. does anyone have one working with asterisk or SER? Sorry, have not used one but I also would be interested in hearing from someone who has. Are these rebadged units from a different manufacturer? Seems likely. Google for voip root wakeup (without the quotes) yields hits from numerous Japanese ISPs that use what appears to be the same box (UI very similar, same case, slightly different indicator functions). For example, see http://www.cypress.ne.jp/web/web-common/setting/voip/connect-adapter/connect-adapter.html The logo on these units is NTT, but I don't know if they are the manufacturer. (An inexpensive unit like this is probably assembled in China, anyhow.) anyone have any experience good or bad with these? --Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Here is the Makefile from asterisk-1.0.0
--- Tomica Crnek [EMAIL PROTECTED] a écrit :
From few days ago there is no USE_MYSQL_FRIENDS in
channels/Makefile. That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed under
the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a écrit :
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of
i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a écrit
:
Hi Harry,
how did you make sip peers on mysql
database? is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails ?
Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
Créez votre Yahoo! Mail sur
http://fr.benefits.yahoo.com/
Le nouveau Yahoo! Messenger est arrivé !
Découvrez
toutes les
nouveautés pour dialoguer instantanément avec
vos
amis. A
télécharger gratuitement sur
http://fr.messenger.yahoo.com
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télécharger gratuitement sur
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=== message truncated ===
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Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour
dialoguer
RE: [Asterisk-Users] Re: SIP peers in MySQL Database
The Makefile isn't gonna help with cvs-head since the code was ripped out. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Monday, October 11, 2004 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database Here is the Makefile from asterisk-1.0.0 --- Tomica Crnek [EMAIL PROTECTED] a écrit : From few days ago there is no USE_MYSQL_FRIENDS in channels/Makefile. That is why I am asking this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Monday, October 11, 2004 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database look at ../channels/Makefile try USE_MYSQL_FRIENDS=1 Harry # # Asterisk -- A telephony toolkit for Linux. # # Makefile for Channel backends (dynamically loaded) # # Copyright (C) 1999, Mark Spencer # # Mark Spencer [EMAIL PROTECTED] # # Edited By Belgarath Aug 28 2004 # Added bare bones ultrasparc-linux support. # # This program is free software, distributed under the terms of # the GNU General Public License # OSARCH=$(shell uname -s) PROC=$(shell uname -m) USE_MYSQL_FRIENDS=0 USE_SIP_MYSQL_FRIENDS=0 --- Tomica Crnek [EMAIL PROTECTED] a écrit : It says To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS., but there is no MYSQL_FRIENDS in channels/Makefile any more. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Monday, October 11, 2004 11:45 AM To: Glynn Condez Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: SIP peers in MySQL Database Hi, Look at: http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers http://www.voip-info.org/wiki-Asterisk+configuration+from+database Is it working well? I don't know because of i'm waiting a reply in order to use sql database for all sip clients from small offices asterisk box with nat context. May I use autocreatepeer in all asterisk sip.conf file with nat=yes in general option ??? [general] dbname= Name of database in your Mysql server dbhost= Hostname of server dbuser= Username in MySQL dbpass= Password for user in MySQL autocreatepeer=yes nat=yes --- -- |Asterisk |-- |nat/firewall box | --- -- | | -- | Internet |-- |nat/firewall box|--Asterisk -- + | SIP peers in |mysql database --- -- |Asterisk |-- |nat/firewall box | --- -- Harry --- Glynn Condez [EMAIL PROTECTED] a écrit : Hi Harry, how did you make sip peers on mysql database? is it working well? where can I find a documentation so I could migrate my Asterisk sip config to use Mysql also. Regards Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez
Re: [Asterisk-Users] Re: SIP peers in MySQL Database
Sorry
I have not look at CVS but I would like somebody help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS
in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed
under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a écrit
:
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a écrit
:
Hi Harry,
how did you make sip peers on mysql database?
is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
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Re: [Asterisk-Users] Can't compile chan_h323 in latest CVS...
On Sat, 2004-10-09 at 02:16, deimios wrote: On Sat, 9 Oct 2004 12:56:44 +0800, Walter Klomp [EMAIL PROTECTED] wrote: Hi, In the latest CVS I am trying to compile chan_h323, but it doesn't want to. chan_h323.c: In function `oh323_call': chan_h323.c:453: error: structure has no member named `callerid' chan_h323.c:455: error: structure has no member named `callerid' chan_h323.c:455: error: structure has no member named `callerid' chan_h323.c: In function `oh323_new': chan_h323.c:756: error: structure has no member named `callerid' make[1]: *** [chan_h323.o] Error 1 ...if I unremark the line #CHANNEL_LIBS+=$(shell [ -f h323/libchanh323.a ] echo chan_h323.so) In channels/Makefile... I have successfully made the channels/h323 with the openh323 and the pwlib... But for some reason asterisk is not making the chan_h323. Am I missing something? (I have asked this question a few days ago and nobody responded, am I alone in this?) ~help~please~ Walter I am having similiar problems, I have searched for an answer but nothing has sprung forward. Cruising the code tonight looking for a fix... If someone has a fix already or information on how to fix it please please let us know. -Regards Deimios Hi I walked that path last week. What I got from the IRC channel (don't exactly remember who) is that the is mayor redo on the caller ID structure, you have to wait until the chan_oh323 guys catch up. Use the cvs from 2004-08-30 that compiles OK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343 2085 Ext 199 Venezuela: +58 212 771 3100 Ext 199 Colombia: +57 1 325 6900 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Can't compile chan_h323 in latest CVS...
Use Asterisk v1-0 and please you're using chan_oh323 NOT chan_h323 they are two totally different channel drivers. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pablo Endres Sent: Monday, October 11, 2004 7:49 AM To: deimios; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can't compile chan_h323 in latest CVS... On Sat, 2004-10-09 at 02:16, deimios wrote: On Sat, 9 Oct 2004 12:56:44 +0800, Walter Klomp [EMAIL PROTECTED] wrote: Hi, In the latest CVS I am trying to compile chan_h323, but it doesn't want to. chan_h323.c: In function `oh323_call': chan_h323.c:453: error: structure has no member named `callerid' chan_h323.c:455: error: structure has no member named `callerid' chan_h323.c:455: error: structure has no member named `callerid' chan_h323.c: In function `oh323_new': chan_h323.c:756: error: structure has no member named `callerid' make[1]: *** [chan_h323.o] Error 1 ...if I unremark the line #CHANNEL_LIBS+=$(shell [ -f h323/libchanh323.a ] echo chan_h323.so) In channels/Makefile... I have successfully made the channels/h323 with the openh323 and the pwlib... But for some reason asterisk is not making the chan_h323. Am I missing something? (I have asked this question a few days ago and nobody responded, am I alone in this?) ~help~please~ Walter I am having similiar problems, I have searched for an answer but nothing has sprung forward. Cruising the code tonight looking for a fix... If someone has a fix already or information on how to fix it please please let us know. -Regards Deimios Hi I walked that path last week. What I got from the IRC channel (don't exactly remember who) is that the is mayor redo on the caller ID structure, you have to wait until the chan_oh323 guys catch up. Use the cvs from 2004-08-30 that compiles OK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343 2085 Ext 199 Venezuela: +58 212 771 3100 Ext 199 Colombia: +57 1 325 6900 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie OT Question - Hardware advise
Hello, in advance I'd like to apologize myself for probably stupid questions which follow, I'm just a newbie to Asterisk: I'd like to use Asterisk as VoIP gateway between two PBXen. Ie: Phone Net 1 | PBX 1 --- TelCo | Asterisk 1 | [VoIP] | Asterisk 2 | PBX 2 --- Telco | Phone Net 2 So the calls from PhoneNet1 to PhoneNet2 would be routed through VoIP. I'm sure that this is not some magic and configuring Asterisk won't be that hard. Of course, if there's some HOWTO to this case, let me know... :-) I've got much more important question: Which HW to use? I'm able to connect PBX to Asterisk using ISDN line - I want the solution to be universal. I do need max. 2 simultaneous channels (for now), one line should be fine then - I'd like to use some simple ISDN card. Problem is, that in ISDN4Linux FAQ there's written: quote 3.1 feature_not: Which ISDN features cannot be offered by isdn4linux? . . Such device-specific ISDN features are, among others: rejection of a waiting call, caller id on/off, ... ^ Should I interpret it that simple ISDN cards supported by I4L doesn't support CLI/CLIP/CLIR? It's necessary for my solution that CLI is present on every end - otherwise the PBX won't be able to decide about routing the call through VoIP. Please advise me piece of hardware that's: suitable for this solution, not expensive (I don't want to buy profi stuff as Zaptels right in the beginning) and working in Europe. Has anybody tried something like this? Thanks a lot --ZK - - ---[ CESKE TELEKOMUNIKACE ]-- - - Zdik Kudrle GSM: +420 604 781 414 HTTP: www.cesketelekomunikace.cz SMTP: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI - Zoom X5v built-in VoIP DSL router
Just thought I would let the list know, as we got our pre release versions today of the new Zoom X5 that supports VoIP. The device comes with an RJ11 phone socket on the back and lets you configure your ADSL router to become a SIP phone (using your existing PSTN phone). Better still, it also allows you to switch the phone between landline and SIP, and does it automatically for incoming calls. No idea what the price of these devices will be when they hit the shops, but setting one up today, if anyone thinks it would be helpful I dont mind doing a little review of the hardware once its tested. Model Number is 5565 Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
ast_data is available at
http://svn.asteriskdocs.org/res_data/
roy
On Oct 11, 2004, at 14:47, harry gaillac wrote:
Sorry
I have not look at CVS but I would like somebody help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a crit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS
in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed
under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a crit
:
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of i'm
waiting a
reply in order to use sql database for all sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
| SIP
peers
in
|mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a crit
:
Hi Harry,
how did you make sip peers on mysql database?
is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
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Re: [Asterisk-Users] nufone config
Can someone post or forward me the relevant sections of their nufone
configs?
I seem to be brainfarting on making it work. All my outbound attempts
end up with results like this:
bebop*CLI iax2 debug
IAX2 Debugging Enabled
bebop*CLI set verbose 9
Verbosity was 0 and is now 9
-- Executing SetCallerID(SIP/710-1980, 9104108307) in new stack
-- Executing Dial(SIP/710-1980,
IAX2/[EMAIL PROTECTED]/19104108307) in new stack
-- Called [EMAIL PROTECTED]/19104108307
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 6ms SCall: 1 DCall: 0 [198.22.67.70:4569]
VERSION : 2
CALLED NUMBER : 19104108307
CALLING NUMBER : 9104108307
LANGUAGE: en
USERNAME: andrewkt
FORMAT : 4
CAPABILITY : 14
ADSICPE : 2
DATE TIME : 155805489
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00016ms SCall: 00170 DCall: 1 [198.22.67.70:4569]
CAUSE : No authority found
bebop*CLI
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00016ms SCall: 1 DCall: 00170 [198.22.67.70:4569]
-- IAX2/nufone/1 is circuit-busy
-- Hungup 'IAX2/nufone/1'
== Everyone is busy/congested at this time
-- Executing Congestion(SIP/710-1980, ) in new stack
== Spawn extension (trusted, 19104108307, 3) exited non-zero on
'SIP/710-1980'
bebop*CLI
I've tried several variations of friend/user/peer in iax.conf but
haven't been able to make anything happen. My most recent config looks
like this one, which is basically a duplicate of my voicepulse connect
entry, which does work.
register = andrewkt:[EMAIL PROTECTED] ; knock knock...
[nufone]
type=friend ; yes, i know friend is evil, it was a last resort attempt
host=switch-2.nufone.net
username=andrewkt
context=default
auth=md5
secret=mypass
At some point last night, I think I had * registering properly with
nufone, as it showed up when I did iax2 show registry Now, it does
not. I'm not worried about that yet, as my (brand new) toll free did
doesn't seem to be working anyway (doesn't ring to failover, I get a
message from my LEC saying the number is disconnected).
My dialout line is a copy of my working voicepulse out section:
[NuFoneOut]
exten = _1NXXNXX,1,SetCallerID(9104108307)
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,3,Congestion
exten = _1NXXNXX,4,Hangup
My context in iax.conf is NANPA or some such. Cannot look at it now.
Although I do remember seeing NANPA in some example configs somewhere,
it didn't change my results.
--
Andrew Thompson
http://aktzero.com/
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Re: [Asterisk-Users] Re: Re[2]: cisco ip 7905 legal ..
Hello Pavel, well .. any GOOD propisition for the same or lower price would be nice IP300 and ip500 are more expensif than this one -- Best regards, Dannymailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. Monday, October 11, 2004, 12:22:45 PM, you wrote: PJ so, better is to look to another phone, than surcharge cisco ;-) PJ PJ PJ - Original Message - PJ From: AST 386sx PJ Newsgroups: gmane.comp.telephony.pbx.asterisk.user PJ Sent: Monday, October 11, 2004 2:00 AM PJ Subject: Re: Re[2]: cisco ip 7905 legal .. PJ If you are going to buy it new. It should not be a problem at all. Just PJ order the SIP software with the phone. PJ Your order should be somthing like this. PJ CP-7905G - Cisco IP Phone 7905G, Global PJ SW-SMH-UL-7905 - SIP or H.323 license for single 7905 IP phone PJ CON-SNT-CP7905 - 8x5xNBD Svc, Cisco IP Phone 7905 PJ Not the following software(s). PJ SW-CCME-UL-7905 - Cisco CallManager Express License For Single 7905 IP Phone PJ SW-CCM-UL-7905 - CallManager Unit license for single 7905 IP phone PJ Both softwares should be same price. I don't think you can change the PJ software license or type of software even you hav smartnet. PJ --ast386-- PJ - Original Message - PJ From: Danny Zak [EMAIL PROTECTED] PJ To: Kannaiyan Natesan [EMAIL PROTECTED] PJ Cc: Asterisk Users Mailing List - Non-Commercial Discussion PJ [EMAIL PROTECTED] PJ Sent: October 10, 2004 3:32 PM PJ Subject: Re[2]: [Asterisk-Users] cisco ip 7905 legal .. Hello Kannaiyan, i need to know the correct procedure; otherwise i will bringing my customers in danger and that is not what i want. i know you can buy the 7905 WITHOUT the callmanager license.. if i load the sip image in it; will that be ok ? PJ ___ PJ Asterisk-Users mailing list PJ [EMAIL PROTECTED] PJ http://lists.digium.com/mailman/listinfo/asterisk-users PJ To UNSUBSCRIBE or update options visit: PJhttp://lists.digium.com/mailman/listinfo/asterisk-users PJ ___ PJ Asterisk-Users mailing list PJ [EMAIL PROTECTED] PJ http://lists.digium.com/mailman/listinfo/asterisk-users PJ To UNSUBSCRIBE or update options visit: PJhttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Yes but it's will be better to have mysql driver At 14:20 11/10/2004, you wrote: Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Hi all,
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
Anybody could answer to my first question ?
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
ast_data is available at
http://svn.asteriskdocs.org/res_data/
roy
On Oct 11, 2004, at 14:47, harry gaillac wrote:
Sorry
I have not look at CVS but I would like somebody
help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a
écrit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS
in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark
Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed
under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a
écrit
:
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any
more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of
i'm
waiting a
reply in order to use sql database for all
sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
|
SIP
peers
in
|
mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a
écrit
:
Hi Harry,
how did you make sip peers on mysql
database?
is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
Vous manquez d'espace pour stocker vos mails
?
Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
Créez votre Yahoo! Mail sur
http://fr.benefits.yahoo.com/
Le nouveau Yahoo! Messenger est arrivé !
Découvrez
toutes les
nouveautés pour dialoguer instantanément avec
vos
amis. A
télécharger gratuitement sur
=== message truncated ===
Vous manquez despace pour stocker vos mails ?
Yahoo! Mail vous offre GRATUITEMENT 100 Mo !
Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/
Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour
dialoguer instantanément avec vos amis. A télécharger gratuitement sur
http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] Newbie OT Question - Hardware advise
caller id on/off, ... ^ Should I interpret it that simple ISDN cards supported by I4L doesn't support CLI/CLIP/CLIR? No, it yust says that you cannot select by software if to transmit caller id. If the line is configured to generayyl transmit ID it should be ok for you. Elmar ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
look at unixODBC or iodbc for more information
Also the reason (i guess) why they move to ODBC is that's ODBC have many
connector to most SQL database.
At 15:26 11/10/2004, you wrote:
Hi all,
Just two questions:
Why asterisk use ODBC(Microsoft?) to connect to SQL
database?
Anybody could answer to my first question ?
Harry
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit
:
res_config_odbc and ast_data is the new way
the old way is still in 1.0.1 and CVS -r v1-0
ast_data is available at
http://svn.asteriskdocs.org/res_data/
roy
On Oct 11, 2004, at 14:47, harry gaillac wrote:
Sorry
I have not look at CVS but I would like somebody
help
me too about my problem.
help please
--- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a
écrit
:
it's in there in -r v1-0, but replaced by some
realtime stuff in
development CVS
I haven't found out more about that, though..
On Oct 11, 2004, at 13:36, Tomica Crnek wrote:
From few days ago there is no USE_MYSQL_FRIENDS
in channels/Makefile.
That is why I am asking this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 12:19 PM
To: Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Re: SIP peers in
MySQL Database
look at ../channels/Makefile
try USE_MYSQL_FRIENDS=1
Harry
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for Channel backends (dynamically
loaded) # #
Copyright (C) 1999, Mark Spencer # # Mark
Spencer
[EMAIL PROTECTED] # # Edited By
Belgarath Aug
28 2004 # Added bare bones ultrasparc-linux
support.
#
# This program is free software, distributed
under the terms
of # the GNU General Public License #
OSARCH=$(shell uname -s)
PROC=$(shell uname -m)
USE_MYSQL_FRIENDS=0
USE_SIP_MYSQL_FRIENDS=0
--- Tomica Crnek [EMAIL PROTECTED] a
écrit
:
It says To enable this, you need to edit the
Makefile in
the channels
directory of your source tree and enable
MYSQL_FRIENDS.,
but there is
no MYSQL_FRIENDS in channels/Makefile any
more.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of
harry gaillac
Sent: Monday, October 11, 2004 11:45 AM
To: Glynn Condez
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: SIP peers in
MySQL
Database
Hi,
Look at:
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
http://www.voip-info.org/wiki-Asterisk+configuration+from+database
Is it working well? I don't know because of
i'm
waiting a
reply in order to use sql database for all
sip
clients from
small offices asterisk box with nat context.
May I use autocreatepeer in all asterisk
sip.conf
file with
nat=yes in general option ???
[general]
dbname= Name of database in your Mysql server
dbhost=
Hostname of server dbuser= Username in MySQL
dbpass= Password
for user in MySQL autocreatepeer=yes nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall
box|--Asterisk
--
+
|
SIP
peers
in
|
mysql
database
--- --
|Asterisk |-- |nat/firewall box |
--- --
Harry
--- Glynn Condez [EMAIL PROTECTED] a
écrit
:
Hi Harry,
how did you make sip peers on mysql
database?
is
it working well?
where can I find a documentation so I could
migrate my Asterisk sip
config to use Mysql also.
Regards
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RE: [Asterisk-Users] Re: SIP peers in MySQL Database
You must be one of those people that doesn't know much about ODBC and is under the impression it's SLOW! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Arnaud Pignard Sent: Monday, October 11, 2004 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database Yes but it's will be better to have mysql driver At 14:20 11/10/2004, you wrote: Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Hi all, hi Just two questions: Why asterisk use ODBC(Microsoft?) to connect to SQL database? To bypass licencing issues in MySQL? Anybody could answer to my first question ? To bypass licencing issues in MySQL? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Why asterisk use ODBC(Microsoft?) to connect to SQL database? 1. It's not Microsoft at all. 2. It's unixODBC (I don't see Microsoft here at all) 3. Wider database support without having to know each database type. 4. It's not much slower than native DB drivers. (15-33% slower) But In my tests you would never see this unless you're doing 10k selects and 5k inserts and that's on a 1ghz box. Anybody could answer to my first question ? Not really sure what your first question is and I'm not gonna dig for it. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
look at unixODBC or iodbc for more information Also the reason (i guess) why they move to ODBC is that's ODBC have many connector to most SQL database. Bingo bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seeking a VoIP Solution for a big company
HI Everybody! My company is seeking to replace its legacy PBX by a VoIP solution; since we prioritize the Open Source Paltform we have found Asterisk doing our own research and we are very interested in it. Knowing that we are decided to make the move to VoIP, can somebody tells me the feasibility of deploying such a solution in an environment that has the following technical requirements: - 250 Users for the Headquarter (100 Mb LAN) - Around 50 remote sites ( WAN Technology: Leased lines/ISDN/VPNADSL/Wireless) - Unified messaging - Small call center (10 users) - CTI Applications - Interoperability with the existing carriers ( Phone companies/ 64 lines) - Security Thanks for your reply _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
[EMAIL PROTECTED] wrote: But In my tests you would never see this unless you're doing 10k selects and 5k inserts and that's on a 1ghz box. Per seconds? Per day? -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Here is my first question.
Two smalls offices with sip clients
+ Asterisk, one offices with Asterisk and mysql
database.
I would like to define all sip peers in mysql database
so Asterisk from small office could read sip peers
configuration from database office.
May I use autocreatepeer in all asterisk sip.conf file
with nat=yes in general option ?
Regards
Harry
[general]
dbname= Name of database in your Mysql server
dbhost= Hostname of server
dbuser= Username in MySQL
dbpass= Password for user in MySQL
autocreatepeer=yes
nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall box|-Asterisk
-- +
| SIPpeers in
| mysql database
--- --
|Asterisk |-- |nat/firewall box |
--- --
--- Brian West [EMAIL PROTECTED] a écrit :
Why asterisk use ODBC(Microsoft?) to connect to
SQL
database?
1. It's not Microsoft at all.
2. It's unixODBC (I don't see Microsoft here at all)
3. Wider database support without having to know
each database type.
4. It's not much slower than native DB drivers.
(15-33% slower)
But In my tests you would never see this unless
you're doing 10k
selects and 5k inserts and that's on a 1ghz box.
Anybody could answer to my first question ?
Not really sure what your first question is and I'm
not gonna dig for it.
bkw
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Re: [Asterisk-Users] newbie question - app_realtime.so failed
Somehow you are out of sync with CVS. app_realtime is not in the 1.0 branch. ast_load_realtime is defined in config.c so somehow you got the soruce to app_realtime but didn't get an updated config.c and many others. If everything is working now, just make a noload=app_realtime.so Matthew - Original Message - From: mihai iancu [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 10, 2004 9:05 PM Subject: [Asterisk-Users] newbie question - app_realtime.so failed Hello, Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18 Everything was running nice and clean with an old version from Aug 2004. Cleaned all source code and binaries - download and install version 1.0 and this is what I get: Oct 10 22:44:36 WARNING[8192]: /usr/lib/asterisk/modules/app_realtime.so: undefined symbol: ast_load_realtime Oct 10 22:44:36 WARNING[8192]: Loading module app_realtime.so failed! Any ideas? Thank you. ___ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: SIP peers in MySQL Database
Per second. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andreas Sikkema Sent: Monday, October 11, 2004 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database [EMAIL PROTECTED] wrote: But In my tests you would never see this unless you're doing 10k selects and 5k inserts and that's on a 1ghz box. Per seconds? Per day? -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Where did USE_SIP_MYSQL_FRIENDS go?
All db specific code has been removed from the code in favor of the currently-in-development RealTime method of configuration from database. You are most likely not using the 1.0 stable branch. You need to use the new RealTime configuration method. And currently, there is only support for odbc. I am currently in the final stages of finishing the RealTime MySQL driver. You can either revert back to 1.0 or wait a few more days for this dev code to be released. Matthew - Original Message - From: Tomica Crnek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 1:33 AM Subject: [Asterisk-Users] Where did USE_SIP_MYSQL_FRIENDS go? There is no USE_MYSQL_FRIENDS and USE_SIP_MYSQL_FRIENDS in .../asterisk/channels/Makefile any more. But, on voip-info wiki it still says that it should be configured like this. Anyone knows how should I tell Asterisk to use mysql database for SIP and IAX friends? Thanks Tomica Crnek ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing calls
Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 onRedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error,
-- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stackOct 11 13:49:12 WARNING[262159]: channel.c:1901 ast_request: No channel type registered for 'Modem'Oct 11 13:49:12 NOTICE[262159]: app_dial.c:742 dial_exec: Unable to create channel of type 'Modem' == Everyone is busy/congested at this timeOct 11 13:49:22 WARNING[262159]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'default'
Extension 2001 gives "unreachable"99 is thecode using for outgoing calls.
;sip.conf[2001]type=friendsecret=2001auth=2001callerid="user 2001" 2001host=dynamicdisallow=allcontext=defaultallow=ulawallow=alaw
;extensions.conf[default]exten = 2001,1,NoOp( call for ${EXTEN})exten = 2001,2,Dial(SIP/${EXTEN},60,tr)exten = 2001,3,Congestionexten = _99.,1,Dial(Modem/ttyI0:${EXTEN:0},20,r)
;modem.conf[interfaces]context=remotedriver=i4llanguage=entype=autodetectdialtype=tonemode=ringdevice = /dev/ttyI0
Have I missed something in my extensions.conf? or in modem.conf?
thanks for your support...
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Re: [Asterisk-Users] Re: SIP peers in MySQL Database
I am in the final stages of writing res_config_mysql.so So far, all of my internal testing with it works. Stand by.. Matthew - Original Message - From: Arnaud Pignard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 7:09 AM Subject: Re: [Asterisk-Users] Re: SIP peers in MySQL Database At 13:53 11/10/2004, Roy Sigurd Karlsbakk wrote: it's in there in -r v1-0, but replaced by some realtime stuff in development CVS I haven't found out more about that, though.. Old mysqlfriends is now remove from asterisk. Now you have to use res_config_odbc for setup sip/iax friends. you can read wiki and this file README.extconfig in docs for get more information how to setup it. You will find also example in extconfig.conf.sample Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. -- Arnaud Pignard ([EMAIL PROTECTED]) Frontier Online - Opérateur Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Intel Modem vs Digium Cards
Cheap shot. Digium does Asterisk FOR FREE. No. As with most of us who support free software projects, we support them because it suits our business goals. We don't do it for free. The investment in time, effort, and resources is paid back, frequently in a way which can't directly be translated by accountants, but it is still an investment, and it is expected to pay off. There are massive benefits to having other users in the community contributing towards and extending the development. Some of us don't even actively *advertise* our company's association with the project in question, something which has been mildly nagging at me about the Digium situation. They support themselves, which I hope you agree is a necessary thing, by selling hardware, one instance of which is the low-end X100P. Essentially the X100P is a slightly modified generic voicemodem THAT COMES WITH CUSTOMER SUPPORT. That is, along with its hardware functionality comes the ability to call up and get help if you encounter problems. That seems quite reasonable. This list is intensely active, and the developers and others who provide advice here are necessarily limited in the amount of attention they can devote to (the often repetitive) questions coming from first-timers. That seems quite reasonable as well. There are, of course, many other participants on the lists, and numerous resources which can be used to help solve problems. Stir into that mix a first-timer who is undercutting the profit model that enables Digium to offer us this wonderful software, And don't forget to trivialize the contributions of everyone else while you're doing it, and then sprinkle your obnoxious insult to the community on top, I didn't find it obnoxious or insulting. In fact, I'd have to agree. One of the benefits to the whole free software movement is supposed to be the freedom to make choices (or, if you prefer, the freedom not to be locked in to a vendor). If you're going to jump all over a guy who *wants* to join the community, for not buying your Approved Vendor's Hardware, maybe because he can't afford it or justify the cost, then it is you who are damaging and limiting the growth of the community. I would imagine that Digium made a conscious choice to use an existing generic voicemodem chipset and to make its drivers compatible with generic versions. As a manufacturer, they certainly had the option to obfuscate things at the hardware level - and they didn't. If they truly wanted to discourage people from doing this, why distribute a driver package that recognizes and installs generic devices? I believe Digium recognizes that they are adding significant value to an otherwise-worth-$2.50-in-quantity, and are betting that most people will see value in buying in at a premium. However, it appears to me that they have also chosen to invite people in who, for whatever reason, have not chosen to purchase their hardware. Looking at it from their point of view, that makes *sense*, because if someone invests five bucks at Fry's on a crummy softmodem, puts it in their box, discovers the joys of Asterisk, and then sells other people on the wonders of Asterisk, Digium still stands to profit. The community grows, and being the main supplier of Asterisk-compatible interface cards should remain a profitable business because most commercial installations will want some level of support. So for heaven's sake, don't dump on some guy for buying a generic softmodem so he can play around. Encourage it. Say generic softmodem is better than alienating this guy. and you're going to find that people (correctly) tell you to go away and solve your own problems. Wow, that's a really sucky attitude. I would expect *Digium* to tell him to go away and solve his own problems. However, if the user community does that, then this is one of the suckiest user communities I've run across in the free software world, and I've been doing free software for many years. From my perspective your primary problem isn't hardware; its your attitude. And from mine, it's users with attitudes like yours. As for me? I'm shopping for cheap modem cards. Why? 1) I'm on FreeBSD, so Digium probably won't support that. 2) I realistically expect to go all VoIP, except perhaps for fax, so I don't want to spend a ton on cards that I won't need. 3) I expect to do something like a Sipura 3000 if we retain a single POTS line, or maybe some sort of Cisco with ISDN BRI VIC cards if we keep the BRI's. 4) I don't really think my PPro200 PBX box will survive very well with having to handle the codec work anyways. But I'm open to spending ten bucks to explore this method. If I was buying a Digium card and it didn't pan out, I'd probably want to see if I could return it, and then there's all the annoyance of an RMA, and time frames after which you can't return it, etc. This way, I'm out a whopping $10.90, and I can deal with that.
RE: [Asterisk-Users] Re: SIP peers in MySQL Database
No , i use unixODBC on several application/servers. but as you said : 4. It's not much slower than native DB drivers. (15-33% slower) I have never done any bench about it. So i can't make any argumentation on it and seems you have done some bench. However add unixODBC on the middle won't be faster. Let's see future usage realtime external, and imagine all configuration (extension ...) in database on busy server. I would prefer have native mysql driver to reduce load than unixODBC. For most asterisk installation, i agree, unixODBC will fit perfectly. At 15:45 11/10/2004, you wrote: You must be one of those people that doesn't know much about ODBC and is under the impression it's SLOW! bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Arnaud Pignard Sent: Monday, October 11, 2004 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: SIP peers in MySQL Database Yes but it's will be better to have mysql driver At 14:20 11/10/2004, you wrote: Somebody seems start a mysql drivers for realtime external configuration instead of ODBC. You can speak to MySQL with ODBC. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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, PSTN, 911, and hardware question
I don't think you want a latching relay, unless you know how to build the support circuit -- a latching relay has two coils and requires a short pulse of power on either coil to change state. The advantage is that it doesn't need any power to hold state, but of course the circuit isn't straightforward anymore. I used: http://www.allelectronics.com/cgi-bin/category.cgi?category=searchitem= RLY-625type=store On mine and hooked it up to an internal 5V supply of the * box. When the box is off, one of my cordless phones gets the line for 911; when the box is on, the cordless is an extension on *, and the PSTN line goes into an X100P. -Original Message- From: Rajeev Sharma [mailto:[EMAIL PROTECTED] Sent: Sunday, October 10, 2004 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Vonage, PSTN, 911, and hardware question OK, first of all, thanks for all the great help everybody. It's nice to see that * has such a nice community! Anyway, that double-pole-double-throw relay looks like just the right thing. If I'm understanding right, the relay design that Henry Devito sent me is the exact same thing as the Viking PF-6A. So, has anyone had experience with these things? Are they easy to build? (This is a home project, so things don't have to be professional.) Any tips? Right now I'm thinking of trying to build something out of this $1.25 12VDC relay (I believe Henry said it had to be 12V): http://www.allelectronics.com/cgi-bin/category.cgi?category=50 0item=RLY-87type=store ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] DID trunk suggestions for Asterisk
Doing some further searching it looks as though as Steve pointed out earlier the TDM400P may work for this. Has anyone else used the TDM400P to handle analog DID trunks? Steve Underwood said: Hi, Technically you can do it, but whether you can get that as a service depends on where you live. This may be what the original poster refered to. Steve harry gaillac wrote: hello, You can do DND over BRI ! Harry --- Joe Cunningham [EMAIL PROTECTED] a écrit : I have a client that has 4 analog DID trunks which are wink start lines and are the incoming lines. Each trunk line has 20 DIDs assigned. We are in the process of migrating them off their current system to Asterisk. We need a way to get these DID trunks into Asterisk so we get the DID information to route the calls. While the DID setup self is not the issue. The issue is I am looking for an inexpensive way to get these trunks directly into the Asterisk box. Since they are analog DID we need to supply the voltage to the CO. I have looked at using something like a BrookTrout card but these are costly. Also looked using something like the DID-200 to convert the DID trunk to loop-start again these have a high cost. I also considered having the client order and switch to PRI and all the advantages this brings however I think this might be overkill for this client. Not sure of the month costs with this I was thinking depending on the term this may run about $500 per month. I also thought about BRI but searching Google and such first it looks like you can't do DID over BRI. So I guess my question is based on the above have I missed something and is there any inexpesive way to bring these DID trunks right into Asterisk keeping the current analog DID trunks? If so what hardware is required? Cheers Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Thank you for your reply. I forgot to mention ... Asterisk dies with that error message ... Everything goes ok with download/compile but when I want to run Asterisk it dies. Message: 7 Date: Sun, 10 Oct 2004 21:14:53 -0500 From: Brian West [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] newbie question - app_realtime.so failed To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Because realtime isn't in 1.0 or 1.0.1 its ONLY in cvs-head. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mihai iancu Sent: Sunday, October 10, 2004 9:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newbie question - app_realtime.so failed Hello, Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18 Everything was running nice and clean with an old version from Aug 2004. Cleaned all source code and binaries - download and install version 1.0 and this is what I get: Oct 10 22:44:36 WARNING[8192]: /usr/lib/asterisk/modules/app_realtime.so: undefined symbol: ast_load_realtime Oct 10 22:44:36 WARNING[8192]: Loading module app_realtime.so failed! Any ideas? Thank you. __ Do you Yahoo!? Y! Messenger - Communicate in real time. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD incomming CALL won't authenticate in SIP
Hi List, I've successfully got Asterisk up and running, can make out going calls fine, It can also register FWD OK, but when a Call comes in from outside it is rejected with this message. Oct 11 09:09:40 NOTICE[98310]: chan_sip.c:7175 handle_request: Failed to authenticate user 499xxxsip:[EMAIL PROTECTED];tag=a41e8548 However if I make a setting in the sip.conf called the FWD incomming number it works fine. It just means I can receive calls unless I put them in my sip.conf like this [499xxx] ;a friends FWD number type=user nat=yes host=dynamic context=fwd-inbound canreinvite=no qualify=yes insecure=yes here is my SIP.conf register = 499yyy:[EMAIL PROTECTED]/499yyy [fwd] ; inbound connections from FWD type=user nat=yes host=dynamic context=fwd-inbound canreinvite=no qualify=yes insecure=yes [fwd-499yyy] ; make outbound calls with this type=friend secret=x username=499yyy host=fwd.pulver.com context=fwd-outbound nat=yes canreinvite=no disallow=all allow=ilbc allow=ulaw What am I doing wrong? BJ _ Smart Saving with ING Direct earn 5.25% p.a. variable rate: http://ad.au.doubleclick.net/clk;7249209;8842331;n?http://www.ingdirect.com.au/burst6offer.asp?id=8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7910 MWI
Hi I cant get the Message waiting indicator to light on my 7910 phones. What am I missing? Here is a snip of my skinny.conf [Guest] device=SEP00044DE12922 version=PC040300 host=192.168.254.18 nat=0 callerid=Henry Devito 1277 mailbox=1277 callwaiting=1 transfer=1 threewaycalling=1 context=general line = 1277 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Grandstream phone price
Wolf N. Paul wrote: Except that £55 is more like $75-80 and not $35. Regards, Wolf Reminds me of a wonderful anecdote about a college english professor who, upon reading in one of his student's compositions that a character had fallen down stairs and laid prostrate on the floor, that the student had failed to make the distinction between a fallen woman and a woman who had merely fallen. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
Greg, Which kernel are you using? I have two machines at home and the zaptel kernel module only runs properly on one of them... The P-3 box worked... kernel-2.4.20-30.9.i686.rpm The Athlon did not... kernel-2.4.20-31.9.athlon.rpm Both machines were updated on the same day (apt-get) and for the most part have the same packages. Has anyone made any headway on this? Thanks, Pete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] System Hang Problem
I am getting some weird behavior and a rash of interesting messages in the log files. If anyone has some ideas, it would be appreciated. Using Asterisk v1.0.1 on Suse Enterprise Linux v8.0. HP DL380 Server. 4GB Ram - Dual 3.2ghz processors. This first entry is when asterisk simply goes unresponsive. We've got a script that automatically polls asterisk (via sip) and restarts it if it does not receive a response. Notice the 9:56 to 10:01 gap. Oct 11 09:53:29 WARNING[6427661]: Failed to write frame Oct 11 09:55:53 WARNING[6445068]: Failed to write frame Oct 11 09:56:10 WARNING[6449163]: Failed to write frame Oct 11 10:01:59 NOTICE[6478861]: Removed default indication country 'us' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:01:59 NOTICE[6150]: Cannot find extension context 'default' Oct 11 10:02:01 NOTICE[1024]: parking.conf is deprecated in favor of 'features.c We've started getting allot of these messages in our log files. Unlikely that this is not associated with the first problem. Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:05 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:05 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to build sip pvt data for MWI Oct 11 10:02:06 WARNING[6150]: Unable to allocate socket: Too many open files Oct 11 10:02:06 WARNING[6150]: Unable to create RTP session: Too many open files ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Intel Modem vs Digium Cards
Why don't you take this off-line were it belongs Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Monday, October 11, 2004 9:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards Cheap shot. Digium does Asterisk FOR FREE. No. As with most of us who support free software projects, we support them because it suits our business goals. We don't do it for free. The investment in time, effort, and resources is paid back, frequently in a way which can't directly be translated by accountants, but it is still an investment, and it is expected to pay off. There are massive benefits to having other users in the community contributing towards and extending the development. Some of us don't even actively *advertise* our company's association with the project in question, something which has been mildly nagging at me about the Digium situation. They support themselves, which I hope you agree is a necessary thing, by selling hardware, one instance of which is the low-end X100P. Essentially the X100P is a slightly modified generic voicemodem THAT COMES WITH CUSTOMER SUPPORT. That is, along with its hardware functionality comes the ability to call up and get help if you encounter problems. That seems quite reasonable. This list is intensely active, and the developers and others who provide advice here are necessarily limited in the amount of attention they can devote to (the often repetitive) questions coming from first-timers. That seems quite reasonable as well. There are, of course, many other participants on the lists, and numerous resources which can be used to help solve problems. Stir into that mix a first-timer who is undercutting the profit model that enables Digium to offer us this wonderful software, And don't forget to trivialize the contributions of everyone else while you're doing it, and then sprinkle your obnoxious insult to the community on top, I didn't find it obnoxious or insulting. In fact, I'd have to agree. One of the benefits to the whole free software movement is supposed to be the freedom to make choices (or, if you prefer, the freedom not to be locked in to a vendor). If you're going to jump all over a guy who *wants* to join the community, for not buying your Approved Vendor's Hardware, maybe because he can't afford it or justify the cost, then it is you who are damaging and limiting the growth of the community. I would imagine that Digium made a conscious choice to use an existing generic voicemodem chipset and to make its drivers compatible with generic versions. As a manufacturer, they certainly had the option to obfuscate things at the hardware level - and they didn't. If they truly wanted to discourage people from doing this, why distribute a driver package that recognizes and installs generic devices? I believe Digium recognizes that they are adding significant value to an otherwise-worth-$2.50-in-quantity, and are betting that most people will see value in buying in at a premium. However, it appears to me that they have also chosen to invite people in who, for whatever reason, have not chosen to purchase their hardware. Looking at it from their point of view, that makes *sense*, because if someone invests five bucks at Fry's on a crummy softmodem, puts it in their box, discovers the joys of Asterisk, and then sells other people on the wonders of Asterisk, Digium still stands to profit. The community grows, and being the main supplier of Asterisk-compatible interface cards should remain a profitable business because most commercial installations will want some level of support. So for heaven's sake, don't dump on some guy for buying a generic softmodem so he can play around. Encourage it. Say generic softmodem is better than alienating this guy. and you're going to find that people (correctly) tell you to go away and solve your own problems. Wow, that's a really sucky attitude. I would expect *Digium* to tell him to go away and solve his own problems. However, if the user community does that, then this is one of the suckiest user communities I've run across in the free software world, and I've been doing free software for many years. From my perspective your primary problem isn't hardware; its your attitude. And from mine, it's users with attitudes like yours. As for me? I'm shopping for cheap modem cards. Why? 1) I'm on FreeBSD, so Digium probably won't support that. 2) I realistically expect to go all VoIP, except perhaps for fax, so I don't want to spend a ton on cards that I won't need. 3) I expect to do something like a Sipura 3000 if we retain a single POTS line, or maybe some sort of Cisco with ISDN BRI VIC cards if we keep the BRI's. 4) I don't really think my PPro200 PBX box will survive very well with having to handle the codec work anyways. But I'm open to spending ten bucks to
Re: [Asterisk-Users] Intel Modem vs Digium Cards
Why don't you take this off-line were it belongs You don't think discussions about the Asterisk user community belong on asterisk-users? It belongs right here. Participants who want to alienate potential new users just because they didn't buy a Digium product have a negative effect on the community, and on Digium. I don't hear any whining about people using Asterisk in an all-VoIP configuration, where Digium also doesn't make any direct profit. Let's just say I heartily disagree with your contention that this belongs off-line. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD incomming CALL won't authenticate in SIP
Beau Walker a écrit : [...] here is my SIP.conf register = 499yyy:[EMAIL PROTECTED]/499yyy [fwd] ; inbound connections from FWD type=user nat=yes host=dynamic context=fwd-inbound canreinvite=no qualify=yes insecure=yes This is not needed. You have type=user and below type=friend (which include user) Anyway, host=fwd.pulver.com and not dynamic [fwd-499yyy] ; make outbound calls with this type=friend secret=x username=499yyy host=fwd.pulver.com context=fwd-outbound nat=yes canreinvite=no disallow=all allow=ilbc allow=ulaw type=peer if [fwd] section stay -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] TTS via text2wave
Donny Kavanagh said: Could these files be cached as well? Not sure what files you're refering to but the AGI Perl script isn't being cached as I've been able to change it and call the extension to see the changes without a reload. No res_perl going on here unless it magically part of the stock build now; don't think so. I don't think the sound files are being cached as their names are pretty unique as generated by the Perl File::Temp module. Is there a way to enable additional debugging of the activity in * due to the STREAM FILE command from my AGI? Doing a set verbose and set debug with really big numbers doen's give me anything useful. Thanks again, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk MGCP and DPX 2203 Cable Modem With MTA
Hi all, We've completed asterisk 1.0.0 and patched it to work with MGCP 1.0 and NCS 1.0, also we've registered the DPX 2203 Cable Modem with embedded MTA and it works fine except: - It can't detect off-hook state until I press flash in phone and, - It wont ring when I dial from another phone (it rings only when I use ring test command in MTA console of modem). Can anybody help me? Thanks , Astrit ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 peers in MySQL Database
hello,
I wrote to [EMAIL PROTECTED] in order to
someone help me without reply ?
May be you could help me
Here is my problem.Two smalls offices with sip clients
+ Asterisk, one offices with Asterisk and mysql
database.
I would like to define all sip peers in mysql database
so Asterisk from small office could read sip peers
configuration from database office.
May I use autocreatepeer in all asterisk sip.conf file
with nat=yes in general option ?
Regards
Harry
[general]
dbname= Name of database in your Mysql server
dbhost= Hostname of server
dbuser= Username in MySQL
dbpass= Password for user in MySQL
autocreatepeer=yes
nat=yes
--- --
|Asterisk |-- |nat/firewall box |
--- --
|
|
--
| Internet |-- |nat/firewall box|-Asterisk
-- +
| SIPpeers in
| mysql database
--- --
|Asterisk |-- |nat/firewall box |
--- --
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Re: [Asterisk-Users] Problems with voice menu
--- ismaelg [EMAIL PROTECTED] wrote: Hello all, I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto,contexto_extensiones exten = 2,1,Goto,contexto_operadora The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = 2,2,Background(itranser/trans_operadora) exten = 2,3,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = 1,1,Background(itranser/msg_pasar_ext) My problem, is when I touch the key 1 in my phone, after the msg_bienvenida, asterisk do not pass me to the correct context [contexto_extensiones]. Asterisk do not pass me to any context, asterisk do nothing when I press the 1 key on my phone. Have I missed something in my extensions.conf? or in sip.conf? I think this exten = 1,1,Goto,contexto_extensiones Should be exten = 1,1,Goto(contexto_extensiones,1,1) Umar Sear ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - all new features - even more fun! http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail attachment volume
I have my asterisk voicemail set up to e-mail me .wav attachments (in the wav49 format), and I receive the messages fine, but the volume is so low that I have to turn my speakers as high as they will go in order to hear it (which makes it interesting if I forget to turn them down immediately after). I have searched the wiki, but I cannot seem to find any information about where to change the record level. Does anyone here have any experience with this problem? Thanks, Ron Frederick Ron, Check http://bugs.digium.com/bug_view_page.php?bug_id=0002023. The problem is known. - Brent Is there a fix/patch that can be applied to allow the voicemails to be recorded LOUDER? I would like to go live with my Asterisk system, but this is a major problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Dial group continues to ring after answer - SNOM phones and solution
Asterisk Users;
Just wanted to let you know I fixed my problem.
To follow up on my own testing of the situation, I find that the
continued ringing after pickup only occured on the SNOM phones in the
group. The Grandstream phones stop ringing when another phone picks up.
Having turned on SIP debugging I have verified that the cancel message
is sent to the SNOM phone (and others in the group) when one of the
group phones is picked up, as expected.
It appears that the SNOMs don't handle the cancel message the same as
the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
latest official release.
It seems that these phones even though they are set to do automatic
update, they do not. Or perhaps that capability was broken in the
firmware version I had last updated to.
THE SOLUTION:
To remedy the problem I upgraded to version 3.52 beta version. Also
2.04g fixes this problem as well.
I had to create my own internal TFTP server and flash update to 3.52.
The standard update process did not work to go beyond 2.03y or 2.04g. I
tried 2.05e f and these would never come out of boot.
MORAL TO THE STORY: Keep your SIP phone firmware up to date.
SNOM support is telling me to upgrade to 3.54. I don't see this one
listed on the standard update URL. I am a little leery about moving to
that one.
Now to upgrade my GrandStream's. They seem to be stuck at an old version
as well.
Thanks,
Mike Meyer
On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
Asterisk Users:
We have our * dial plan set up to ring 5 phones in the office and it
delivers the call to the first that picks up their receiver.
The problem is that after the pickup, everyone else's SIP phone keeps
ringing at least once and sometimes twice. This interferes with the
conversation, while others pick up the phone and get nothing.
Does anyone else have similar problems or have a solution to stop the
ring once answered? My dial statement looks like the following and has a
timeout of 15 seconds.
exten = MainTeam,1,Dial(${MainTeamChannels},15,tr)
exten = MainTeam,2,Voicemail(u${MainTeam_EXT})
...
note the variables MainTeamChannels define the SIP phone channels
defined in sip.conf and MainTeam_EXT is the voicemail box for this group
extension.
As an alternative, I am considering doing a round robin on a call group
or pickup group and implementing call pickup.
Any ideas welcome.
Mike Meyer
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[Asterisk-Users] SOHO small or rack mount chassis and mobo for asterisk
What is anyone out there using that's small, quiet and robust for a SOHO system with two X100P and a TDM400? I'd love to see some recos for easy to find hardware to build asterisk office pbx. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reading global vars from AGI
is there any way to read global vars like ${EXTEN}, ${GROUPCOUNT} from an
AGI?
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RE: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution
Someone pointed me here
http://www.snom.com/downloads/share (had to guess at URL as the Snom
site appears to be down or uber slow but if that's not it its damn close
:-P )
Which lists all versions of firmware for all their phones. Handy if you
have a specific version in mind but don't know the correct URL. Tho I
haven't had problems with the auto-update so far.
HTH
alex
-Original Message-
From: Mike Meyer [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 16:12
To: Asterisk Users Group
Subject: [Asterisk-Users] Re: Dial group continues to ring after answer
-SNOM phones and solution
Asterisk Users;
Just wanted to let you know I fixed my problem.
To follow up on my own testing of the situation, I find that the
continued ringing after pickup only occured on the SNOM phones in the
group. The Grandstream phones stop ringing when another phone picks up.
Having turned on SIP debugging I have verified that the cancel message
is sent to the SNOM phone (and others in the group) when one of the
group phones is picked up, as expected.
It appears that the SNOMs don't handle the cancel message the same as
the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
latest official release.
It seems that these phones even though they are set to do automatic
update, they do not. Or perhaps that capability was broken in the
firmware version I had last updated to.
THE SOLUTION:
To remedy the problem I upgraded to version 3.52 beta version. Also
2.04g fixes this problem as well.
I had to create my own internal TFTP server and flash update to 3.52.
The standard update process did not work to go beyond 2.03y or 2.04g. I
tried 2.05e f and these would never come out of boot.
MORAL TO THE STORY: Keep your SIP phone firmware up to date.
SNOM support is telling me to upgrade to 3.54. I don't see this one
listed on the standard update URL. I am a little leery about moving to
that one.
Now to upgrade my GrandStream's. They seem to be stuck at an old version
as well.
Thanks,
Mike Meyer
On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
Asterisk Users:
We have our * dial plan set up to ring 5 phones in the office
and it
delivers the call to the first that picks up their receiver.
The problem is that after the pickup, everyone else's SIP phone
keeps
ringing at least once and sometimes twice. This interferes with the
conversation, while others pick up the phone and get nothing.
Does anyone else have similar problems or have a solution to
stop the
ring once answered? My dial statement looks like the following and has
a timeout of 15 seconds.
exten = MainTeam,1,Dial(${MainTeamChannels},15,tr)
exten = MainTeam,2,Voicemail(u${MainTeam_EXT})
...
note the variables MainTeamChannels define the SIP phone channels
defined in sip.conf and MainTeam_EXT is the voicemail box for this
group extension.
As an alternative, I am considering doing a round robin on a
call
group or pickup group and implementing call pickup.
Any ideas welcome.
Mike Meyer
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Dear Friends of Ubiquity Software:
As you may have noticed, Ubiquity Software began using the web domain ubiquity.com
earlier this year in addition to the previously established ubiquity.net for our
website and email communications to you. However, since that time, a dispute has
emerged with respect to actual ownership of the ubiquity.com domain.
As an international software company founded over decade ago, you can always reach
Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and
via email at @ubiquity.net. However, we have also chosen to expand our domain to the
more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and
@ubiquitysoftware.com for email communications.
Please use either the historical ubiquity.net or begin to use the new
ubiquitysoftware.com domain for all email communications to Ubiquity employees from
now on.
Thank you.
Regards,
Ubiquity Software
www.ubiquitysoftware.com http://www.ubiquitysoftware.com/
[EMAIL PROTECTED]
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Re: [Asterisk-Users] SIP peers in MySQL Database
You have obviously never posted to any kind of mailing list before. Sometimes you may have to wait a few days for someone to answer you. Sometimes people just don't know. Griping to the owners of the list about the people who take time out of their day to give you FREE support isn't going to make things better nor will it make you popular nor will you get a faster response (if any). As long as the database can be accessed by the asterisk server, then you can store sip info into that database. You should not need to use autocreate. If you don't need immediate, uptodate, realtime sip configuration, look in ASTERISK SOURCE ROOT/contrib/scripts/ for something called retreive_sip_from_mysql.pl or something like that. That is what I use. Or you can be patient and the new RealTime method should be in stable form in a week or two. Matthew - Original Message - From: harry gaillac [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Monday, October 11, 2004 10:01 AM Subject: [Asterisk-Users] SIP peers in MySQL Database hello, I wrote to [EMAIL PROTECTED] in order to someone help me without reply ? May be you could help me Here is my problem.Two smalls offices with sip clients + Asterisk, one offices with Asterisk and mysql database. I would like to define all sip peers in mysql database so Asterisk from small office could read sip peers configuration from database office. May I use autocreatepeer in all asterisk sip.conf file with nat=yes in general option ? Regards Harry [general] dbname= Name of database in your Mysql server dbhost= Hostname of server dbuser= Username in MySQL dbpass= Password for user in MySQL autocreatepeer=yes nat=yes --- -- |Asterisk |-- |nat/firewall box | --- -- | | -- | Internet |-- |nat/firewall box|-Asterisk -- + | SIPpeers in | mysql database --- -- |Asterisk |-- |nat/firewall box | --- -- Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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-sccp2
How do you install this? I downloaded it from sourceforge, but I can not find a documentation or how-to ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Access Bank II
It says FXS. How are you setting your switches? All the rear panel switches are set to normal but Im unsure of the front. Thanks, Mason Herring From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim McKee Sent: Sunday, October 10, 2004 8:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Access Bank II might want to check you port card and be sure you have fsx ports rather than fxo ports... I'm running a II with stock configs and having no problems. tim mckee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mason Herring Sent: Sunday, October 10, 2004 3:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Access Bank II I show an active channel in zttool when taking a line off-hook but * provides no dial tone. Any guesses? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mason Herring Sent: Saturday, October 09, 2004 9:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Access Bank II Heres the zaptel configs. Ztcfg shows no alarms. Thanks for the help! Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXO Loopstart (Default) (Slaves: 03) Channel 04: FXO Loopstart (Default) (Slaves: 04) Channel 05: FXO Loopstart (Default) (Slaves: 05) Channel 06: FXO Loopstart (Default) (Slaves: 06) Channel 07: FXO Loopstart (Default) (Slaves: 07) Channel 08: FXO Loopstart (Default) (Slaves: 08) Channel 09: FXO Loopstart (Default) (Slaves: 09) Channel 10: FXO Loopstart (Default) (Slaves: 10) Channel 11: FXO Loopstart (Default) (Slaves: 11) Channel 12: FXO Loopstart (Default) (Slaves: 12) Channel 13: FXO Loopstart (Default) (Slaves: 13) Channel 14: FXO Loopstart (Default) (Slaves: 14) Channel 15: FXO Loopstart (Default) (Slaves: 15) Channel 16: FXO Loopstart (Default) (Slaves: 16) Channel 17: FXO Loopstart (Default) (Slaves: 17) Channel 18: FXO Loopstart (Default) (Slaves: 18) Channel 19: FXO Loopstart (Default) (Slaves: 19) Channel 20: FXO Loopstart (Default) (Slaves: 20) Channel 21: FXO Loopstart (Default) (Slaves: 21) Channel 22: FXO Loopstart (Default) (Slaves: 22) Channel 23: FXO Loopstart (Default) (Slaves: 23) Channel 24: FXO Loopstart (Default) (Slaves: 24) Channel 25: FXO Loopstart (Default) (Slaves: 25) Channel 26: FXO Loopstart (Default) (Slaves: 26) Channel 27: FXO Loopstart (Default) (Slaves: 27) Channel 28: FXO Loopstart (Default) (Slaves: 28) Channel 29: FXO Loopstart (Default) (Slaves: 29) Channel 30: FXO Loopstart (Default) (Slaves: 30) Channel 31: FXO Loopstart (Default) (Slaves: 31) Channel 32: FXO Loopstart (Default) (Slaves: 32) Channel 33: FXO Loopstart (Default) (Slaves: 33) Channel 34: FXO Loopstart (Default) (Slaves: 34) Channel 35: FXO Loopstart (Default) (Slaves: 35) Channel 36: FXO Loopstart (Default) (Slaves: 36) Channel 37: FXO Loopstart (Default) (Slaves: 37) Channel 38: FXO Loopstart (Default) (Slaves: 38) Channel 39: FXO Loopstart (Default) (Slaves: 39) Channel 40: FXO Loopstart (Default) (Slaves: 40) Channel 41: FXO Loopstart (Default) (Slaves: 41) Channel 42: FXO Loopstart (Default) (Slaves: 42) Channel 43: FXO Loopstart (Default) (Slaves: 43) Channel 44: FXO Loopstart (Default) (Slaves: 44) Channel 45: FXO Loopstart (Default) (Slaves: 45) Channel 46: FXO Loopstart (Default) (Slaves: 46) Channel 47: FXO Loopstart (Default) (Slaves: 47) Channel 48: FXO Loopstart (Default) (Slaves: 48) 48 channels configured. --zapata.conf-- [channels] ; ; Default language ; ;language=en ; ; Default context ; context=default signalling=fxo_ls group=1 channel = 1-48 --/etc/zaptel.conf-- span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs fxols=1-48 loadzone = us defaultzone=us From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Saturday, October 09, 2004 8:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Access Bank II Can you post your configs please? At least the parts that pertain to your question. It may help to figure out if there is a problem. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mason Herring Sent: Saturday, October 09, 2004 8:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Access Bank II I have a CAC Access Bank II with 2 T1 and 48 analog ports and am using an Asterisk server with 2 T100P cards. Both Access Bank and T100 cards have green lights. Zaptel.conf and Zapata.conf are configured for fxo loop start and I show 48 configured channels when doing ztcfg. When we take an analog line from one of the 48 ports
Re: [Asterisk-Users] chan-sccp2
Henry Devito ([EMAIL PROTECTED]) wrote: How do you install this? I downloaded it from sourceforge, but I can not find a documentation or how-to currently i am writing one ... --jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] RTP timing issues
Dear Sirs, The Asterisk bounty has been updated accordingly. Some info about our environment: Our Asterisk server is logically connected to a Veraz NGN platform through SIP and we are facing two major problems for calls from/to Veraz; When calling from Veraz to any SIP extension, no ringback is generated as Veraz does not generate any RTP packets until Answer supervision. Asterisk can not deliver ringback. Calling to Veraz is problematic as all our interfaces are using Silence compression. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, October 07, 2004 11:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; Bart Coppens Subject: Re: [Asterisk-Users] RTP timing issues On Thu, 7 Oct 2004, Bart Coppens wrote: Some time ago, I announced a bounty to solve the issues with regards to silence compression (chopped voice) and one way voice. To get this solved, Asterisk should get the clocking from an internal source in a way that an ouput stream can be generated without getting any RTP input. Now my company is exposing a payment of 1000USD for this bounty. This payment have to justified through an official invoice. Can someone give me an indication if this can be achieved? It can be achieved. Steve _ All about Paris Motor Show 2004 http://motorshow.auto.msn.be ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-sccp2
Thanks, I will be patient and wait. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Czmok Sent: Monday, October 11, 2004 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan-sccp2 Henry Devito ([EMAIL PROTECTED]) wrote: How do you install this? I downloaded it from sourceforge, but I can not find a documentation or how-to currently i am writing one ... --jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P to Verizon Smart Jack Question
Has anybody had any experience connecting the t100p to a verizon smart jack. I've been told the t100p uses an RJ48 but not the revision (i.e. C, S, X ) I've created wires (RJ48C x-over) but no green light on the t100p 1-4 2-5 4-1 5-2 i've created wire (RJ48S) no green light (only because the HyperEdge Smart Jack says this is default but can't confirm this to be true in my case). 1-7 2-8 7-1 8-2 Standard RJ45 ether cable gets ligts on the card and smart jack but no signal zttool indicates OK for both the t100p and the tdm04b cards (no alarms) I can ping the t100p but no further with the standard ether cable I'm at a loss Regards Greg Cirino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] windows messenger
is it possible to windows messenger clients of an asterisk server to chat (text chat) with each other? what about the status presence? is it possible to each windows messenger client of an asterisk server to see the presence on other clients? if not, what is missing in asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] windows messenger
Asterisk doesn't support MSN9 the protocol Windows Messenger (and MSN Messenger) uses to communicate with a messenger server such as MSN or Windows 2003 running the Live Conferencing server. It should be possible to write an MSN9 server independently of Asterisk since the information needed by such a server is available via the Manager API. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shabanip Sent: October 11, 2004 4:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] windows messenger is it possible to windows messenger clients of an asterisk server to chat (text chat) with each other? what about the status presence? is it possible to each windows messenger client of an asterisk server to see the presence on other clients? if not, what is missing in asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] support
Hi, I'm try to get asterisk up and runing on my linux pc, but I can't download the file (asterisk,zaptel libpri), i got connect to your ftp server but I can't download the files from asterisk or diguim, i login as anonymous, i saw the pub file but i can't got it,if somebody give a hand to acomplish my linux pbx project. Thanks You!! Rock, jazz, country, soul & more. Find the music you love on MSN Music! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail attachment volume
Is there a fix/patch that can be applied to allow the voicemails to be recorded LOUDER? I would like to go live with my Asterisk system, but this is a major problem. Its not asterisk that's the problem I suspect. If you get low recordings you need to look into using app_test to help find them. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
