similar to: Using Asterisk's manager interface to recieve calls

Displaying 20 results from an estimated 5000 matches similar to: "Using Asterisk's manager interface to recieve calls"

2008 Jan 04
3
Using Asterisc for Taking Calls for Radio
Hello Asterisc-Users List, I am new to the list. I joined with a question in mind: How would you set up an asterisc box so that: (A) Someone dials a number (B) They are presented with a menu (C) Entering a number, like 1, connects a call to me. (D) I am on a mixing board, running an internet radio show. I want to run asterisc into the board, and run an output from the board to asterisc. Is that
2007 Mar 09
1
sip tunnel
Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.
2007 Mar 13
1
voicemail scenario
Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance.
2007 Jul 18
2
what codecs for LAN
Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel --------------------------------- Don't pick lemons. See all the new 2007 cars at
2007 Jun 18
1
atxfer attended transfer feature
I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works
2007 May 08
1
hardened kernel and nut access to ttyS
Hi, I am running nut with megatec driver accessing ttyS0 as user nut on "standard" kernel (gentoo-sources). It works fine. However, I just built a hardened kernel on a new gentoo machine and have no experience with it. NUT (upsdrv) is failing because it says it doesn't have permission to access ttyS0 even though nut is within the appropriate group. I can add user = root in ups.conf
2007 May 30
0
Call transfer while dialing
Hi, I want to transfer the call to a conferencing room while dialing. I tried to do that using manager API(Redirect), but it did't work. Regards, Jason. ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html
2007 Jun 23
1
Zaptel Compilation Error
Hi List; I think my problem in Zaptel compilation is related to autoconf: no input file, anyone has an advise? Also, I did a change in the Makefile existed in the following path: /usr/src/kernels/2.6.20-1.2319.fc5-i686/ EXTRAVERSION = 2.6.20-1.2319.fc5 Now, if I run uname -r then I get output: 2.6.20-1.2319.fc5 But the directory under the kernels is: 2.6.20-1.2319.fc5-i686 So do I have to
2007 Jan 16
1
Refreshing DNS lookups
Hi there The "dnsmgr" in Aterisk 1.4.0 seems not to work. I enabled "DNS lookups" in dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups. Any ideas how to debug this issue? Thanks in advance Housi Mueller --------------------------------- Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. -------------- next part
2007 Jun 07
1
call Hold event asterisk
i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is
2010 Jul 26
0
cli_session_setup_blob: recieve failed (NT_STATUS_INVALID_PARAMETER)
Hi, I'm trying to access a share on my work network using smbclient. We have an Windows Active Directory network. My client computer is running Solaris 10 u8. The computer hosting the share says it's running Acopia ARX(3.0.0b1) According to Active Directory (not familiar with this OS, i think it is a NAS) I run this command to get the Kerberos ticket. bash-3.00$ kinit jtmb at
2009 Oct 29
0
automatically recieve mail
Hi! I want to make mobile phone registration system with e-mail confirmation. firstly, When user send the e-mail from the cell phone then user will recieve an register url by e-mail. It likes an error handling system, when error happen the system will send the error message but it is a bit different I think. how can i do it? please give me a clue. -- Posted via http://www.ruby-forum.com/.
2010 Apr 26
1
Building Asterisk-RPM for 1.4.24.1
Hi everybody, quite frequently I build customized RPMs with asterisk-1.4.20.1 including some special patches for it, to install the on CentOS 5. Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is not working anymore with my build environement. In version 1.4.22 the "Makefile" was modified and all the RPM-stuff was removed, same for the
2007 May 08
2
Dovecot Startup error
Hi, I have installed Dovecot 1.0.0 on a FreeBSD6.0 machine with Exim 4.66 and Vexim 1.5. When I restart the machine, dovecot does not load properly. The logs of /var/log/maillog are as: May 8 00:10:32 lhr dovecot: pop3-login: No authentication sockets found May 8 00:10:32 lhr dovecot: child 11478 (login) returned error 89 May 8 00:10:34 lhr dovecot: imap-login: No authentication sockets
2004 Nov 25
2
How to make/recieve call using asterisk when thereis a power failure?
Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange
2006 Apr 04
1
Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.
I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk and
2007 Jan 23
7
access users homes share
hey list, we are currently migrating our users from novell to samba. now we have one problem: in novell we could give e.g. user1 access to users2 home share so he could modify, delete or add files on this share. in samba we defined a global homes share that is mapped on logon. so how can we give user1 the needed rights? here is the definition of the homes share in smb.conf: [homes]
2013 Nov 14
1
recieve fax from PRI using spandsp 65% failure rate
Hi, On one of our servers, we're having problems with incoming faxes. The connections come in through a PRI into a Digium TE820 card. 'fax show stats' yields the following: FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 740 Completed FAXes : 740 Failed FAXes : 486 Spandsp G.711
2006 Apr 04
0
Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafax
I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfully but I have problems with some fax machine so I wanted to try using HylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my problem. I'm trying to connect Asterisk with SpanDSP using iaxmodem. My system looks like this: ISDN <---> Asterisk <---> IAXModem <---> Hylafax Asterisk
2007 Jul 16
1
[Asterisk]Asterisk's behavior of a simple call
Hello, I tried to configure a very simple case of Asterisk using SIP userA --- Asterisk server ---- userB sip.conf [userA] type=friend username=userA host=dynamic nat=no context=test [userB] type=friend username=userB host=dynamic nat=no context=test In extensions.conf [test] exten => 1000,1,Dial(SIP/userA) exten => 2000,1,Dial(SIP/userB) I make a call from userA to userB, it works,