[asterisk-users] Asterisk 1.4 Hint is not detected the extensions status
Dear Friends and Supporters! I try to install the Asterisk 1.4, and I needs to activate the hint to for the call pickup feature. However, the hint is enabled and I can see the status of the extensions by run command show hints. It show the phones are Idle. However, it would NOT be able to detect the extensions if they are ringing or Inuse at all, that why it did not sent the ringnotify to the subscribed phone. When the phones are ringing or in talking, the show hints still show the phones are Idle. Any ideal? I would very appreciated for your help. Lan. ___ --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 realtime with mysql 5.0 and unixODBC.
Dear Friends and Supporters! I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0 and unixODBC to the beta asterisk 1.4. I run the make and make install for the asterisk-addon just fine, It created the modules res_config_mysql.so and cdr_addon_mysql.so without any problem or error. However, when I run the asterisk, it comes up with the error : == Parsing '/etc/asterisk/res_mysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: mysql_init and the asterisk would not be run. However, if I do the noload those modules noload = res_config_mysql.so noload = cdr_addon_mysql.so Then the asterisk running just fine, but there is no database connection for asterisk realtime. Would anyone help me, I would very appreciated. Thanks in advance! Lan. ___ --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: Asterisk Manager
Dear Michiel and Supporters!
Thank you for your reply!
Here is the complete code for my monitor.php
html
head
titleAsterisk Status/title
link href=common/classic.css rel=stylesheet type=text/css
/head
body
h2Asterisk Status: ? echo `hostname`; echo (.$SERVER_ADDR.); ?/h2
?
$arr = array(
Uptime = asterisk -r -x 'show uptime',
Database Connection Status = asterisk -r -x 'realtime mysql
status',
Active Channel(s) = asterisk -r -x 'sip show channels',
Working Queues = asterisk -r -x 'show queues',
Registered Phones = asterisk -r -x 'sip show peers',
Zaptel driver info = asterisk -r -x 'zap show channels',
);
foreach ($arr as $key = $value) {
?
br
table class=box
tr class=boxheader
td class=boxheader? echo $key; ?/td
/tr
tr class=boxbody
td
table border=0
tr
td
pre? echo passthru($value); ?/pre
/font/td
/tr
/table
/td
/tr
/table
?
}
?
/body
/html
# End of monitor.php file ##
The manager.conf as I have posted previously.
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 25, 2006 1:37 PM
Subject: Re: [asterisk-users] Re: Asterisk Manager
On 13:12, Wed 25 Oct 06, Maps wrote:
Dear Friends and Supporters!
I try to write a php application to monitor the asterisk, but when I
open the .php to access to asterisk to retrieve the information about the
queues status, sip show peers, realtime mysql status etc... However, It
just return to me Unable to connect to remote asterisk (does
/var/run/asterisk.ctl exist?)
Asterisk is current running with the a file in /var/run/asterisk.ctl for
the user asterisk. I have set asterisk to be the owner of the folder
/var/run, and start asterisk with user is asterisk. HTTPD is run under
asterisk user. My manager.conf has an entry.
[admin]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
However, my php still unable to retrieve the information for asterisk.
Did I miss somethings?
How are you connecting to asterisk?
Maybe you can paste some code so we can actually see why it
is not working.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] Asterisk Manager
Dear Friends and Supporters! I try to write a php application to monitor the asterisk, but when I open the .php to access to asterisk to retrieve the information about the queues status, sip show peers, realtime mysql status etc... However, It just return to me "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My manager.conf has an entry. [admin]secret = passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,userwrite = system,call,log,verbose,command,agent,user However, my php still unable to retrieve the information for asterisk. Did I miss somethings? Your help would be very appreciated! Regards, Lan ___ --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 Manager
Dear Friends and Supporters! I try to write a php application to monitor the asterisk, but when I open the .php to access to asterisk to retrieve the information about the queues status, sip show peers, realtime mysql status etc... However, It just return to me "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My manager.conf has an entry. [admin]secret = passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,userwrite = system,call,log,verbose,command,agent,user However, my php still unable to retrieve the information for asterisk. Did I miss somethings? Your help would be very appreciated! Regards, Lan ___ --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] Webphone with Asterisk??
Dear Friends and Supporters! Could we be able to use the webphone with asterisk? If we can, could you please tell me how or where could I find the information? Thanks! Lan ___ --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 Upgrade.
Dear Friends and Supporters! Would anyone tell me how to do the upgrade for asterisk? Thanks for your help! Lan ___ --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] BroadVoice subscribers and Asterisk 1.2.3
It's not only with the * 1.2.3. I am still using the 1.1 with [EMAIL PROTECTED] 1.5, but I also have the same problem with BV. Lan - Original Message - From: Ronald Lewis To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 11:58 AM Subject: [Asterisk-Users] BroadVoice subscribers and Asterisk 1.2.3 I just upgraded a box to 1.2.3 this morning after encountering the issues noted earlier on the list. Everything is great. In fact, a LOT better.In the past few weeks, I've been battling with BV to address dropped outgoing voice packets (the flipside is that I haven't experienced this with other providers during tests), and an annoying mechnical 'chirp' at the start of a call. Since 1.2.3, I haven't (so far) noticed anything unusual.Regards,Ronald LewisFounder CTA, RiverscapeIndependent ConsultantDenver, Colorado303-557-0153[EMAIL PROTECTED][EMAIL PROTECTED]www.riverscapecorp.comwww.ronaldlewis.com -- Listen to my recent interview with Digium's Mark Spencer at http://www.ronaldlewis.com/coffee ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2?
Dear Charles Wang and all Asterisk users and supporters! Thank Charles for giving me this intruction and it works greate for me after I copy the into the modules directory. I didn't know if that easy. However, after I copy and did the reload command, it didn't work until I have to do the restart now. Again, Thank Charles! and hope you guys enjoy the new version of asterisk. Regards, Lan. - Original Message - From: Charles Wang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 22, 2006 6:02 AM Subject: Re: [Asterisk-Users] Installing the none commercial intel g729codecsinto [EMAIL PROTECTED] 2.2? I have the same problem too. I install the G.729 (IPP) to asterisk 1.0.x, and it works well. When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine. I can use show translation and find it too. But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line allow=g729 from sip.conf. And asterisk works fine. 2006/1/22, Guillermo Salas M [EMAIL PROTECTED]: Con fecha 21/1/2006, Francesco Peeters (Asterisk) [EMAIL PROTECTED] escribió: On Sat, January 21, 2006 23:21, Franz Bräuer said: Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I'm using g723.1 and works very well. Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? Kewl! Those work like a treat! As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: cd /usr/lib/asterisk/modules/ wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so After reloading, 'show translation' gives: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 -22 8 817 8 724 115 19897 gsm 151 - 7 716 7 623 114 19796 ulaw 14616 - 111 2 118 109 19291 alaw 14616 1 -11 2 118 109 19291 g726 154241010 -10 926 117 20099 adpcm 14616 2 211 - 118 109 19291 slin 14515 1 110 1 -17 108 19190 lpc10 161311717261716 - 124 207 106 g729 16939252534252441 - 215 114 speex 16030161625161532 123 - 105 ilbc 17343292938292845 136 219 - Jolly good show, old chap! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ --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 -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2billing Trunk
Thanks for this information! Lan - Original Message - From: ram To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 16, 2005 8:50 PM Subject: Re: [Asterisk-Users] A2billing Trunk Hi check this URL will help you for this since this list only asterisk.. not [EMAIL PROTECTED] http://sourceforge.net/forum/forum.php?thread_id=1398290forum_id=420324 ram On 12/17/05, snacktime [EMAIL PROTECTED] wrote: On 12/16/05, MapsAir [EMAIL PROTECTED] wrote: Dear helpers and supporters! I have been playing with the A2billing for a week, but I still get stuck on creating a working trunk to terminate the call.Is any one show me how to setup a trunk in A2billing or pointing me to some where that I can find out the information.As Kevin has already said before, this is off topic and should betaken elsewhere.I think it would be a good idea to create a set of rules for the listand post them once a week or so.It's not really fair to expect a new user to have to search through old messages to find a posting from amonth ago about what is or isn't acceptable, even if it is commonsense.If on signup the rules were part of the signup email, and thenreposted to the list every week or so, then there really wouldn't be any excuse for these off topic posts.I would bet that would cut down a lot on the number of off topic posts.Chris___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Register to Asterisk using MAC address.
Dear Supporters! Does any one know how to set the asterisk to allow the phone to register to asterisk using the MAC address? Thanks! Lan Phan. ___ --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] Asterisk and a SPA3000 behind NAT peerregistration
Dear Richard and supporters! I see that you guys could be able to setup the SPA 3000 to connect to the asterisk thru the NAT. I don't know how would to do this. As my understand is that the SPA 3000 is just able to configure with the SIP that not NAT aware in Asterisk. I am trying to configure the SPA 3000 NAT - Internet NAT Asterisk, but I am unsuccess to do that. Also I have the SPA2100 and try to do the same thing. If you guys could be able to do so. Would you PLEASE tell me how could I do that? or please direct me to the information where I can find out the way to setup that up! I am very appreciated Thanks in advance! Lan Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I manually change the host for the peer for the SPA PSTN Line, then everything works fine, and I can make local calls through it. BUT when my router's IP changes... I am lost. (EVEN while the extension SPAN Line 1, or the incomming trunk are registered and DO work.) According to what I read, if I setup the outgoing trunk to type=peer with host=dynamic, then I can make the peer to register itself into the Asterisk, so the asterisk will know where to contact the peer. But I cannot figure out how to make the SPA register as the peer? I can make it register as a type=friend for the incomming traffic, but not as a peer.. Don't bother with the peer. Define both the Line1 and PSTN ports on the SPA to register with asterisk using different UserID/Passwords for each, use port 5060 for one and 5061 for the other, and in asterisk's sip.conf file define them as type=friend. It does work just fine that way. ___ --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
[Asterisk-Users] Question about setup Grandstream HandyTone 488 SIP with Astersik to Travel throught NAT.
Dear Supporters! Is it posible to setup the Grandstream HandyTone 488 SIP ATA FXO FXS to travel throught that NAT to connect to Asterisk inside the other NAT? The setup is HandyTone 488 --NATInternet--NAT-Asterisk. If it works, what type of NAT should I use, and how can i configure it. Thank you in advance for your helps. Ian. ___ --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
[Asterisk-Users] Remote Provisioning for the PA1688 phones.
Dear helpers! I am looking for the software and information about remote provisioning for the PA1688 phones to upload configuration files using TFTP, FTP, or HTTP. Do you guys know how and where I could find those information! I would be very appreciated for your help! Regards, Ian. ___ --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
[Asterisk-Users] IAX phone not hear the other phone ring when calling
Dear Helpers! I have setup my asterisk with the iax phone. However, when I try to dial out to the pstn phone from the iax phone, I can't hear the other phone ring until they pickup the phone. Doesn't anyone know the issue? Doesn't anyone has experience with the problem? Please helping me or why and how I could correct the problem? I would very appreciated! Regards, Lan ___ 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
