Hi,<br><br>I'm new to Asterisk and I would like your views on my beginning of solution.<br>What I wanted to acheive is for Asterisk to be a simple incoming call queue before transferring to specific phones which are not in the scope of asterisk, but registered to the IPBX<br>
<br>My current constraints by the IPBX:<br>- I can only receive one call per line and can only place one outgoing call per line (the second call is possible but the caller will be in hold and will hear an other hold music...)<br>
- IPBX does not let me place SIP Invite on behalf of the caller to the member of the queue, it's ok using "fromuser"<br>- When using "fromuser" for outgoing sip peers, phones registered to the IPBX does not display the name of the original caller (default behavior of asterisk when using fromuser) because when receiving a call, the phones does an ldap request to get the caller name based on the number)<br>
<br>What I tried to do and it seems to be working (with my current configuration, I have Asterisk 1.6.2.9-2+squeeze4):<br>- Each incoming call is redirected to a queue "simplequeue" with dynamic member which login and logout by calling a phone number <br>
- the phone call is passed to the agent via a local channel using Dial to call the agent with different outgoing lines (depending if the outgoing line is used)<br><br>First, Am i in the right way with this solution ? This was the only solution in my mind but I don't know much about Asterisk possibilities :)<br>
This solution seems to be working except that the display of the incoming caller id on the agent is the outgoing line of asterisk and not the original caller id. <br>To bypass this, instead of using Dial inside the local channel, I tried to use the "Transfer" command to do a SIP REFER and the display should works correctly on my system. <br>
<br>Using Transfer() inside a local channel seems to be not working, i tried to use "sip set debug on" but i can't see any sip messages transmitted after the "Transfer" line...<br><br>Thanks for your help.<br>
Regards,<br><br><br>I have the following configuration:<br><br><sip.conf><br><br>[general]<br>...<br># 5 incoming lines<br>register => <a href="http://8301@192.168.1.14/in">8301@192.168.1.14/in</a><br>register => <a href="http://8314@192.168.1.14/in">8314@192.168.1.14/in</a><br>
register => <a href="http://8327@192.168.1.14/in">8327@192.168.1.14/in</a><br>register => <a href="http://8334@192.168.1.14/in">8334@192.168.1.14/in</a><br>register => <a href="http://8341@192.168.1.14/in">8341@192.168.1.14/in</a><br>
# 3 outgoing lines<br>register => <a href="http://8348@192.168.1.14/out">8348@192.168.1.14/out</a><br>register => <a href="http://8349@192.168.1.14/out">8349@192.168.1.14/out</a><br>register => <a href="http://8366@192.168.1.14/out">8366@192.168.1.14/out</a><br>
#login/logout<br>register => <a href="http://8350@192.168.1.14/login">8350@192.168.1.14/login</a><br>register => <a href="http://8351@192.168.1.14/logout">8351@192.168.1.14/logout</a><br>defaultexpiry=3600<br>disallow=all<br>
allow=alaw<br><br>[8348]<br>type=peer<br>host=192.168.1.14<br>fromuser=8348<br>fromdomain=192.168.1.14<br>disallow=all<br>allow=alaw<br>call-limit = 1<br><br>[8366]<br>type=peer<br>host=192.168.1.14<br>fromuser=8366<br>fromdomain=192.168.1.14<br>
disallow=all<br>allow=alaw<br>call-limit = 1<br><br>[8349]<br>type=peer<br>host=192.168.1.14<br>fromdomain=192.168.1.14<br>fromuser=8349<br>disallow=all<br>allow=alaw<br>call-limit = 1<br><br><extensions.conf><br><br>
[default]<br>exten => in,1,Wait(1) ; Wait a second, just for fun<br>exten => in,n,Answer ; Answer the line<br>exten => in,n,Queue(simplequeue,tT);<br>exten => in,n(end),Hangup<br>
<br>exten => login,1,Answer<br>exten => login,n,AddQueueMember(simplequeue,Local/${CALLERID(num)}@support/n,1,)<br>exten => login,n,Playback(agent-loginok)<br>exten => login,n,Hangup<br><br>exten => logout,1,Answer<br>
exten => logout,n,RemoveQueueMember(simplequeue,Local/${CALLERID(num)}@support/n,1,)<br>exten => logout,n,Playback(agent-loggedoff)<br>exten => logout,n,Hangup<br><br>[support]<br>exten => _[A-Za-z0-9].,1, Dial(SIP/${EXTEN}@8349)<br>
exten => _[A-Za-z0-9].,n, Dial(SIP/${EXTEN}@8348)<br>exten => _[A-Za-z0-9].,n, Dial(SIP/${EXTEN}@8366)<br><br><queues.conf><br><br>[simplequeue]<br>musicclass=rhapsody<br>strategy=random<br><br><br><br><br>