similar to: SJphone on PDA registering with Asterisk???

Displaying 20 results from an estimated 9000 matches similar to: "SJphone on PDA registering with Asterisk???"

2005 Jul 17
2
DNS SRV
I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>
2005 May 26
5
SIP Soft Video phone for Asterisk usage
I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald
2005 Jan 12
5
Grandstream Bugetone 101 & mwi
I tried to use message waiting indicator, by "Subscribe for MWI" in the web menu of the phone. However, it does not light up / flash, even if a voice mail is waiting. Where is the switch to turn it to? bye Ronald
2005 May 17
4
Is SKYPE a threat or should we do something (together)
Skype is very succesfsfull and get more and more users, ... we can ignore them, accept them or do something,... My suggestion is that we try to do something, ... If we would peer to each other, than we get soon also a great amount of users together, and than our service becomes more valuable, ... Let's discuss advantages and disadvantages! bye Ronald -- Ronald Wiplinger (CEO of
2005 Jan 18
3
Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)
I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the
2005 May 31
4
Extension context question
I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>
2006 Apr 10
2
Outbound calls through Broadvoice
Hi all, a noob here, I am trying to get outbound calls through asterisk working with Broadvoice. I have consulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register =>
2005 Mar 15
2
Flashpannel: How to get more than 28 buttons?
I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. The description says you can have a hundred buttons, .... Can I have multiple flash pannels? E.g. for each department? bye Ronald
2005 Aug 18
2
asterick and festival...Help!
Earlier this afternoon I had this working exten => 2890,1,Answer exten => 2890,2,GoTo(12) exten => 2890,12,Wait(1) exten => 2890,13,Festival('I can say numbers like') exten => 2890,14,SayNumber(1230001,f) exten => 2890,15,Wait(1) exten => 2890,16,HangUp I was so very proud of myself... All of a sudden after a reboot.... I get the following from the same call plan
2005 Jan 04
3
Where to start. {Scanned}
Hello All, Yep I'm a newbe. I'm just started to play with asterisk. What I have Redhat Fedora Core 2 (New install) 3 X100P cards. I installed zaptel-1.0.3 libpri-1.0.3 asterisk-1.0.3 Where should I start?? -- Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please
2005 Feb 20
1
making ASTCC web page secure ???
How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password, however since it is a cgi script that does not seem to be posible. Any ideas, any suggestions ?
2005 Jan 06
1
calling with out registration
hi, i am using Asterisk CVS-05/31/04. i have the problem that sip clients can make calls over asterisk without registering befor. the xlite is not loged in with any username/secret bit still can make calls over asterisk. how can that be? thx for help. thomas
2005 Mar 23
1
VoiceMail Outgoing Calls and Disconnects
Hey, I currently have the voicemail set up so I can dial out from a voicemail box. I seem to not be able to place them. I have put the number in with and without a 1, and then hit pound and it says "Please wait while I connect your call" and then disconnects. Any ideas as to why?
2005 Jun 17
2
ASTCC Rate Calculation
Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback.
2005 Mar 18
2
Asterisk 1.0.7 Released
Hello everyone, Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now been released. Libpri and -addons have not changed, but have been updated anyway to keep the version numbers consistent. All of the tarballs are available on the ftp site. ftp://ftp.asterisk.org/pub/asterisk/ I have posted the ChangeLogs for easy viewing at the following address.
2005 Jun 01
1
Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?
Over the past 2 weeks I have been able to compile and get an asterisk system up & running on a debian Linux box. I have setup 5 internal sip clients on the lan and all works great! I can also call from outside (PSTN) into the system and reach extensions and services no problem. All is up & running behind a nat firewall with proper ports forwarded and locked down on each device to work
2005 Jun 07
4
I want to move the MySQL server out to another machine
I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers: 0.7269 or 0.2929 ??? bye Ronald Wiplinger
2005 Mar 24
1
realtime - unable to find key
ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`