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<DIV>hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan.</DIV>
<DIV> </DIV>
<DIV>sip.conf</DIV>
<DIV> </DIV>
<DIV>[general]<BR>port = 5060<BR>bindaddr = 0.0.0.0<BR>context = others</DIV>
<DIV>[2000]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2001]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2002]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2003]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV> </DIV>
<DIV>[2004]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2005]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV> </DIV>
<DIV>[2006]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2007]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV>[2008]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2009]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2010]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>############################################################################################</DIV>
<DIV>vi /etc/asterisk/extensions.conf</DIV>
<DIV>[from-zaptel]<BR>exten => s,1,wait(2)<BR>exten => s,n,dial(sip/2000)<BR>exten => s,n,dial(sip/2001)<BR>exten => s,n,Playback(tt-weasels)</DIV>
<DIV>[others]<BR>include => internal<BR>include => outside</DIV>
<DIV>[inside]<BR>exten => _20XX,1,Dial(SIP/${EXTEN})<BR>exten => _20XX,n,VoiceMail(<A href="mailto:${EXTEN}@others,u">${EXTEN}@others,u</A>)<BR>exten => _20XX,n,Hangup()</DIV>
<DIV>[outside]<BR>exten => 2001,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2001,n,Hangup</DIV>
<DIV>exten => 2002,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2002,n,Hangup</DIV>
<DIV>exten => 2003,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2003,n,Hangup</DIV>
<DIV>exten => 2004,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2004,n,Hangup</DIV>
<DIV>exten => 2005,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2005,n,Hangup</DIV>
<DIV><BR>this is the log when i am calling from exten 2000 to outside</DIV>
<DIV><BR>Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)<BR>Verbosity is at least 3<BR>[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call from '2002' to extension '919369613616' <STRONG>rejected because extension not found</STRONG>.</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>any help n support will be highly appreciated<BR>--- On <B>Sat, 13/2/10, Loris Santamaria <I><loris@lgs.com.ve></I></B> wrote:<BR></DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"><BR>From: Loris Santamaria <loris@lgs.com.ve><BR>Subject: [asterisk-users] Call Pickup with 1.6.2.1 and Snom<BR>To: asterisk-users@lists.digium.com<BR>Date: Saturday, 13 February, 2010, 8:39 AM<BR><BR>
<DIV class=plainMail>Hi,<BR><BR>I've used various patches with asterisk 1.4 to have support for call<BR>pickup and notification with good results.<BR><BR>Now I'm trying vanilla 1.6.2 with its official support for "dialog-info<BR>+xml" notifications with no success. This is what i'm doing:<BR><BR>- Phone A has a key configured as type "extension" pointing to Phone B.<BR>- In sip.conf I added notifycid=ignore-context<BR>- Phone A and B and C are in the same callgroup and pickupgroup<BR>- Phone A and B and C are in the same context<BR><BR>Phone C calls Phone B and asterisk generates a notification for phone A:<BR><BR><?xml version="1.0"?><BR><dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:35505@10.40.23.179"><BR><dialog id="35505" call-id="pickup-3c26701519b8-5xxapzoav2u4" direction="recipient"><BR><remote><BR><identity display="Lab
1">sip:35501@10.40.23.179</identity><BR><target uri="sip:35501@10.40.23.179"/><BR></remote><BR><local><BR><identity>sip:35505@10.40.23.179</identity><BR><target uri="sip:35505@10.40.23.179"/><BR></local><BR><state>early</state><BR></dialog><BR></dialog-info><BR><BR>With this notification, Phone A shows on the screen that Phone C is<BR>calling Phone B, and the function key blinks. If one presses the<BR>blinking function key, the phone generates an Invite with replaces, to<BR>try to pickup the call:<BR><BR>INVITE sip:35501@10.40.23.179 SIP/2.0<BR>Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport<BR>From: "Lab 4" <sip:35504@10.40.23.179>;tag=o28fq65rfu<BR>To: "Lab 1" <sip:35501@10.40.23.179><BR>Call-ID: 3c2672b3f35a-dpd0zv11yegl<BR>CSeq: 1 INVITE<BR>Max-Forwards: 70<BR>Contact: <sip:35504@10.40.24.175:5060>;flow-id=1<BR>Replaces:
pickup-3c26701519b8-5xxapzoav2u4<BR>P-Key-Flags: keys="3"<BR>User-Agent: snom320/7.1.39<BR>Accept: application/sdp<BR>Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO<BR>Allow-Events: talk, hold, refer, call-info<BR>Supported: timer, 100rel, replaces, from-change<BR>Session-Expires: 600;refresher=uas<BR>Min-SE: 90<BR>Content-Type: application/sdp<BR>Content-Length: 368<BR><BR>Then asterisk receives the pickup request:<BR><BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: <no from<BR>tag> Totag: <no to tag><BR>[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35505@RedEdelca<BR>[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT)<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl<BR>[Feb 11 10:44:13] VERBOSE[4659]
chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl<BR>[...]<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4]<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: <no from<BR>tag> Totag: <no to tag><BR>[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35505@RedEdelca<BR>[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT)<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl<BR>[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl<BR>[...]<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35505@PICKUPMARK)<BR>[Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - state 2 (In
use)<BR>[Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2'<BR>[Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found for 35505.<BR>[Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-0000000f'<BR>[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-0000000f, SIP callid 3c2672b3f35a-dpd0zv11yegl<BR><BR>After this obviously phone A hasn't picked up the call, and Phone B<BR>keeps ringing.<BR><BR>Did I miss something in the dialplan or is it a bug?<BR><BR>-- <BR>Loris Santamaria linux user #70506 xmpp:<A href="http://in.mc954.mail.yahoo.com/mc/compose?to=loris@lgs.com.ve" ymailto="mailto:loris@lgs.com.ve">loris@lgs.com.ve</A><BR>Links Global Services, C.A. <A href="http://www.lgs.com.ve/" target=_blank>http://www.lgs.com.ve</A><BR>Tel: 0286 952.06.87 Cel: 0414 095.00.10 sip:<A
href="http://in.mc954.mail.yahoo.com/mc/compose?to=103@lgs.com.ve" ymailto="mailto:103@lgs.com.ve">103@lgs.com.ve</A><BR>------------------------------------------------------------<BR>-O9 -omg-optimize -fomit-instructions<BR><BR><BR><BR>-- <BR>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></DIV></BLOCKQUOTE></td></tr></table><br>
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