Do you have multiple devices registering with the 10x extentions? Or is it just the one device?<br><br>Basically, the phone is not sending the correct Caller-ID, for some reason. Whatever caller-id the phone sends, is what will be sent.
<br><br><div><span class="gmail_quote">On 3/28/07, <b class="gmail_sendername">Drew Gibson</b> <<a href="mailto:drew@oanda.com">drew@oanda.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I have some phones (and an ATA) that are shared between two users who<br>each have separate voicemail but they are not behaving as desired nor<br>expected.<br><br>Incoming calls show up on the correct lines.<br>Calls originating from the device are seen, at the terminating device,
<br>as coming from the account listed last in sip.conf, regardless of the<br>line selected.<br><br>This creates three main issues I would like to resolve:-<br>1. The person called sees the wrong callerid<br>2. The CDR records the call against the wrong account
<br>3. Picking up voicemail requires multiple extra steps<br><br>Is there a way around this??<br><br>Scenario:-<br>Phone 1 has three lines 101, 102, 103<br>Phone 2 has 1 line 202<br><br>User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2)
<br>User 2 at Phone 2 sees call coming from extension 103 instead of 101<br><br>With 'sip debug' enabled at the console, I see an INVITE issued (on the<br>Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the
<br>call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202.<br>103 happens to be the last listed in sip.conf and the first listed in<br>'sip show peers' (I have confirmed that this is dependent on the order
<br>in the conf file, not numeric order)<br><br>sip.conf :-<br>[general]<br>port = 5060<br>bindaddr = <a href="http://0.0.0.0">0.0.0.0</a><br>pedantic = no<br>autocreatepeer = no<br>context = sip<br>registertimeout=20<br>
localnet = <a href="http://10.10.10.0/255.255.255.0">10.10.10.0/255.255.255.0</a><br>srvlookup = yes<br>tos=0xb8<br>rtptimeout=300<br>rtpholdtimeout=1800<br>maxexpirey=3600<br>defaultexpirey=1200<br><br>[sip-101]<br>; Aastra 480i phones for general office
<br>type=peer<br>insecure=very<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>host=dynamic<br>dtmfmode=auto<br>canreinvite=no<br>context=office-dial<br>qualify=yes<br>username=101<br>secret=xxxxxx<br>mailbox=101<br>callerid="User 1" <101>
<br><br><br>sip show peers :-<br>103/103 <a href="http://10.10.10.181">10.10.10.181</a> D 5060 OK (157 ms)<br>102/102 <a href="http://10.10.10.181">10.10.10.181</a> D 5060 OK (159 ms)
<br>202/202 <a href="http://10.10.10.184">10.10.10.184</a> D 5060 OK (4 ms)<br>101/101 <a href="http://10.10.10.181">10.10.10.181</a> D 5060 OK (160 ms)
<br><br><br>Asterisk 1.2.15<br>Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA<br><br>--<br>Drew Gibson<br><br>Systems Administrator<br>OANDA Corporation<br>416-593-6767 x322<br><a href="http://www.oanda.com">
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