similar to: Call transfer while dialing

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

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.

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.

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

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

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

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]

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

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

What i need is to recieve a call in a console! I mean i can call from CLI...but can i recieve calls too? If this is possible how is the console identificated and where! Actually i need to call from one Asterisc server console to another(i know what is asterisc server for, but this is a specific task)! Thanks! --------------------------------- Don't pick lemons. See all the new 2007 cars

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

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

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",

I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXXXXXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am

Hi, This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the "New York" time? What value I have to give to GMT offset value? Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

Hi all, We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2 FXO). We want to do a transfer "blind" and "attended" from a line external connected to one FXO. We have made configuration, and transfers from internal lines (FXS) work fine but from (FXO) not. We have made 2 test, one work fine from FXS and the other form FXO no. Test 1, work fine: 1) A

Hi Friends, I installed and configured Asterisk. I am getting my voice mail to my email as attachments. Well. We can check our voice mail by dialing *98. But, I want to check my voice mails by dialing our DID number from a outside telephone. How can I do this? Please help me. Look forward to your response. Thank you. Regards, Chandra. --------------------------------- Ahhh...imagining

Hi List, I'm setting up a system with one TDM400P (2*FXO + 2 * FXS) and one Junghanns QuadBRI on a Fedora Core 6 (Kernel 2.6.20-1.2944.fc6). I'm using the bristuff-0.3.0-PRE-1y-e kit. It download zaptel-1.2.16, libpri-1.2.4 and asterisk-1.2.17 When it's the time for ztcfg to do its job it complains with <ZT_SPANCONFIG failed on span 2: No such device or address (6)>