Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
uname -a Linux xxx 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:37:32 EDT 2006 i686 i686 i386 GNU/Linux modinfo zaptel | grep ^version: version:1.4.0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
I can run zaptel 1.4 normally in other machine on the same OS, only can't run it on 2850. It hangs the OS.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] web-meetme cbmysql not registered
HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the extension which invoke cbmysql, a warning appears: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module and in my start log, it shows [Jan 30 18:40:15] VERBOSE[6702] logger.c: == Parsing '/etc/asterisk/cbmysql.conf': [Jan 30 18:40:15] VERBOSE[6702] logger.c: Found [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got hostname of localhost [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got port of 3306 [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got sock file of /var/lib/mysql/mysql.sock [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got user of root [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got dbname of meetme [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got password of [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: Using Database for Admin User Options [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Connference Application of MeetMe [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMySQL: got Conference Count Application of MeetMeCount [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: CBMuSQL: Early Alert set to 300 seconds. [Jan 30 18:40:15] DEBUG[6702] app_cbmysql.c: Successfully connected to MySQL database. this seems it was loaded successful. what's the matter?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: How to config rtp packetization in 1.4?
Hi, any one test rtp packetization in 1.4?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so
HI, I use fc6 , the latest stable asterisk 1.4, zaptel 1.4 and libpri 1.4 after I installed zaptel and libpri. I can start zaptel. and my te410p card got green lamp. but when I continue to compile and install asterisk, I can't find chan_zap.so compiled. and in my asterisk cli. I can't 'help zap'. that got nothing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I disable send e-mail feature in the voicemail application?
HI, all Can I disable send e-mail feature in the voicemail application?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk + Huawei
need debug * and Huawei, not * and client___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Huawei
Can you give more debug information. Usually codec incorrupt can cause failure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] real time billing system
a2billing___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Dose this trunk do just like IAX2 trunk, to reduce bandwidth?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP and RTP packetization in 1.4
Okay, Thank you. So packetization is a feature of RTP and can work with all of the codecs, isn't it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Netstat Output
I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Are you use digium card? digium pri card offen cause many problems, check zttest___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 register refuse but Dial cmd works!
Hi, I just set two asterisk connect with iax2 trunk.
B server
[user1]
type=user
trunk=yes
context=from-trunk
username=user1
auth=plaintext
secret=passwd
notransfer=yes
A server
register = user1:[EMAIL PROTECTED]
I notice on A's CLI, it shows Registration of 'user1' rejected:
'Registration Refused' from: 'x.x.x.x'. I also use iax2 show registry it say
Unregistered.
But when I use dial cmd: Dial(iax2/user1:[EMAIL PROTECTED]/${exten},30,), I can
call the extension normally.
What's wrong? Why can't I register to B server?
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[asterisk-users] iax2 trunk call limits
Hi, all Can I limit calls in one iax2 trunk just like sip peers do? How? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to setup multiple iax2 trunks between two asterisk server?
Hi, all How to setup multiple iax2 trunks between two asterisk server? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 trunking
I can take 30 calls in one trunk with good voice quality more calls cause awesome sounds___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 trunk voice quality: how many calls cause jitter?
Hi, I use IAX2 trunk between two asterisk server. At a few calls (less than 30) enviorment, both caller and callee hear each other clearly. But when calls reach 45 or above, the quality of sounds is bad. I wonder if a IAX2 Trunk should limit concurrent calls? I use ILBC codec in the trunk. any idea? btw: I use asterisk-1.2.12, zaptel-1.2.8___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2 warning!
I always got this warning after I'm using IAX2 channels . Sep 11 21:44:09 WARNING[30229]: chan_iax2.c:6536 socket_read: Received trunked frame before first full voice frame What's it mean?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: the sounds quality of IAX2 channels are notgood as SIP channels?
I use the latest version of zaptel, asterisk___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] the sounds quality of IAX2 channels are not good as SIP channels?
I use both IAX2 channels and SIP channels. IAX2 channels reduce bandwidth effectively. But sometime my cli show NOTICE[1281]: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock WARNING[1281]: chan_iax2.c:708 jb_warning_output: Resyncing the jb. last_delay 28, this delay 1227, threshold 1062, new offset -1227 WARNING[1281]: chan_iax2.c:6532 socket_read: Received trunked frame before first full voice frame And when concurrent calls are more than 30, IAX2 channels' sounds quality become poor. I use codec ILBC both in SIP channel and IAX2 channel.Sounds always good when I use SIP protocol. this is some of my configuration: jitterbuffer=yes forcejitterbuffer=no ;dropcount=2 ;maxjitterbuffer=1000 ;maxjitterinterps=10 ;resyncthreshold=1000 ;maxexcessbuffer=80 ;minexcessbuffer=10 ;jittershrinkrate=1 trunkfreq=30 trunktimestamps=yes tos=reliability codecpriority=host It's same on both IAX2 side. Any idea?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:sip giving problems, please help.
Yes, I also get these problems occasionally Sep 4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 60 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LT, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 120 ^ Sep 4 17:52:15 WARNING[1597]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:49 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11346 sipsock_read: We could NOT get the channel lock for SIP/gw-442744f0! Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Sep 4 18:50:51 ERROR[1290]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD!___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newzealand zaptel DTMF problem
Hi, all. I'm using TE411P card and asterisk in Newzealand as a VOIP gateway. At the begining, all works fine. I have 50 concurrent calls in busy time. But recently I found some users can't send DTMFcorrectly to my gateway. I found some of them sent less DTMF digits than they acturely dialed, while some others sent more DTMFs than they dialed. eg. one person dials 1234 while my gateway receives 12334. what's wrong? Of cource, I have already turned dtmf relax on. And I use Read cmd in my dial plan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P and SPANDSP
TE405p and spandsp works good. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P and SPANDSP
Hi, All Does any one has successful experience use te410p and spandsp together? Could they work well with all 120 channels receive/send fax at the same time? My practice is that rxfax always get broken fax page. Help!___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P and SPANDSP
It looks like spandsp doesn't fit a busy fax server solution, usually this kind of solution is depending on some onborad dsp card like eicon diva server. But it still a good tool for faxing, and we can expect its t.38 function. Maybe a big surprise. I have some experience using the TE410p and spandsp. I think my max concurrent rxfax has been about 16 or so on a single E1 on a single cpu 3.0Ghz P4. I expect that it could handle at least 30 concurrent faxes, thereafter I think that the disk controller may start to impact performance. My experience of txfax is that it is very sensitive to system load. When I was testing things I was able to get no more than three concurrent instances of txfax going on a system that was otherwise idle. If I was receiving a fax at all via rxfax then I could reliably have no more than one instance of txfax. Craig - Original Message - From: Ma Zhiyong [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 24, 2005 9:21 AM Subject: [Asterisk-Users] TE410P and SPANDSP Hi, All Does any one has successful experience use te410p and spandsp together? Could they work well with all 120 channels receive/send fax at the same time? My practice is that rxfax always get broken fax page. Help! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P + SPANDSP fax problem
Hi, Steve Thank you for your hard work. Yes, I use EuroISDN. My four E1s connect toan Alcatel S12 Switch that works on PSTN. My Telco. turn CRC4 off, sodo I. And I'll askthem turn it on. I have no X-Windows problem. I run * on Redhat 9.0 and my run level is 3. That doesn't tell us very much, unless you tell us where these 4 E1s connect to. Should CRC4 really be switched off? I assume you are using EuroISDN, since you have CCS enabled. That usually works with CRC4 on, although some telcos do the wrong (read flaky) thing and turn CRC4 off.As well as clock sourcing, another thing causing data slips on many machines is using X-windows. Even on really fast machines the rapid interrupts from the frame buffer used for X-windows causes data loss.Regards,Steve___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P + SPANDSP fax problem
I'm sure use my telco as clock src, and my libtiff is also v3.5.7, while problem still exist. Shall I contact with my telco for timing? in zaptel.conf, I set span=1,1,0,ccs,hdb3span=2,2,0,ccs,hdb3span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 To trace rxfax, just turn on debug trace level. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P + SPANDSP fax problem
Hi,
I just setup a fax server by spandsp. But
it doesn't look good. Becauseeach fax I received from my fax machine is
not completed.
I use te410p work with it. While the voice
call is good.
Any ideas?
Trace shows that the fax is received
successfully.
Aug 17 12:01:10 VERBOSE[19571]: --
Executing RxFAX("Zap/94-1", "/var/spool/asterisk/FAX/1124251267.284.tif") in
new stackAug 17 12:01:46 DEBUG[19571]:
==Aug
17 12:01:46 DEBUG[19571]: Pages transferred: 1Aug 17 12:01:46
DEBUG[19571]: Image size: 1728 x
355Aug 17 12:01:46 DEBUG[19571]: Image resolution 7700 x
3850Aug 17 12:01:46 DEBUG[19571]: Transfer
Rate: 9600Aug 17 12:01:46 DEBUG[19571]: Bad
rows 66Aug
17 12:01:46 DEBUG[19571]: Longest bad row run 22Aug 17 12:01:46
DEBUG[19571]: Compression type 2Aug 17 12:01:46
DEBUG[19571]: Image size (bytes) 0Aug 17 12:01:46 DEBUG[19571]:
==Aug
17 12:01:49 DEBUG[19571]:
==Aug
17 12:01:49 DEBUG[19571]: Fax successfully received.Aug 17 12:01:49
DEBUG[19571]: Remote station id: xxAug 17 12:01:49 DEBUG[19571]:
Local station id: Aug 17 12:01:49 DEBUG[19571]: Pages transferred:
1Aug 17 12:01:49 DEBUG[19571]: Image resolution: 7700 x 3850Aug 17
12:01:49 DEBUG[19571]: Transfer Rate: 9600Aug 17
12:01:49 DEBUG[19571]:
==Aug
17 12:01:51 VERBOSE[2999]: -- Channel 0/1, span 4 got
hangup
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[Asterisk-Users] te405p and dell poweredge
Hi, I want to install * and te405p on Dell Poweredge 1850. Can I do that successfully? Any one has successful experience on that scenario? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp configuration
Hi, I receive fax using spandsp. It works, however the tif file it stored has no good quality. Any method to configure that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium card and fax
Hi, I practiced fax transmission over digium card and spandsp. I send the A4 page from TDM FXS port and received it by spandsp. Whole page was transmitted and stored in a tif file. When I open the file, I get a horizontal A4 page and theimage on the pagewas flatted. Any one has idea? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
It sure works when I createthe diretroy
"/var/spool/asterisk/fax/2201001/".HoweverI found it coundn't received
the whole fax which eicon card sent.
That is, when eicon card send a page with 30 lines text,
spandsp only received18 lines. Then hung up.
-- Goto
(macro-faxreceive,s,7) -- Executing RxFAX("Zap/123-1",
"/var/spool/asterisk/fax/2201001/1115688509.76.tif") in new
stack -- Channel 0/30, span 4 got
hangup May 10 09:28:58 DEBUG[6561]: app_rxfax.c:246
rxfax_exec: Got hangup May 10 09:28:58 DEBUG[6561]:
app_macro.c:172 macro_exec: Extension s, priority 7 returned normally even
though call was hung up
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[Asterisk-Users] spandsp
Hi,
I installed spandsp and test it with Eicon card.
When fax begin from eicon card to spandsp. It fails and
shows:
-- Executing RxFAX("Zap/124-1",
"/var/spool/asterisk/fax/2201001/1115604630.5.tif") in new
stack -- Channel 0/31, span 4 got
hangup May 9 10:10:41 DEBUG[4967]: app_rxfax.c:246
rxfax_exec: Got hangup
Any one has this experiment?
How can I get more log infomation about
softmodem?
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[Asterisk-Users] CID Number problem
Hi, all. I'm glad I put asterisk and hylafax
togetherjust like PSTN-Asterisk-Hylafax-Email.And the
fax2email functionworks well.
But I also find some bugs about CID number.
I use TE405P as gateway and Eicon PRI card as fax
card.
When I receive the caller number from PSTN, I found it was
51863500. While I dial the FAX trunk, FaxGetty get the caller number
051863500.
-- Executing NoOp("Zap/124-1", "51863500") in new stack--
Executing Dial("Zap/1-1", "ZAP/g1/51863507") in new stack
Apr 30 13:30:50faxserver FaxGetty[28254]: --
[33:RING CID: 051863500 DAD: 51863507]
Any idea?
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[Asterisk-Users] ast-rad-acc.pl problem
Hi, All I installed a PortaOne's Radius client for my asterisk Server. But I can't run ast-rad-acc.pl after installation. It says "Can't call method "val" on an undefined value at ./ast-rad-acc.pl line 293." It also show "Config file error" in the log file. Has any one meet this problem? Does that mean my perl module problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P and Fax Server
Hi, All Can I use a TE410P card to make a FAX server? Did anybody have some experiences to construct PBX and Fax Server in one box? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
