[asterisk-dev] callerid broken in trunk?
Ch. Benco
bencokakao at gmail.com
Wed Aug 23 09:55:30 MST 2006
hello!
this should probably go to mantis, however, my question can probably
be answered more general so i don't want to open a bugreport for this:
In my setup i have openser working as registrar and calls are sent to
asterisk which only works as a gateway. however, the callerid of the
phones(as registered with ser) are not recognized by asterisk anymore.
also setting them manually in the dialplan is not recognized. is this
simply broken in the current state of trunk or have i missed some major
change?
i've tried this with several revisions > 40722, last one was 40897...
plz see the attached files sip_debug.txt, dialplan.txt and
verbose_log.txt for further information
thanks
chris
-------------- next part --------------
exten => _X.,1, NoOp(${CALLERIDNUM})
exten => _X.,n, Set(CALLERID(all)="Testname" <555123456>)
exten => _X.,n, NoOp(${CALLERIDNUM})
-------------- next part --------------
-- Executing [6564564564 at from-internal:1] NoOp("SIP/10.1.99.153-081c93e0", "") in new stack
-- Executing [6564564564 at from-internal:2] Set("SIP/10.1.99.153-081c93e0", "CALLERID(all)="Testname" <555123456>") in new stack
-- Executing [6564564564 at from-internal:3] NoOp("SIP/10.1.99.153-081c93e0", "") in new stack
== Auto fallthrough, channel 'SIP/10.1.99.153-081c93e0' status is 'UNKNOWN'
-------------- next part --------------
interface: eth0 (10.1.0.0/255.255.0.0)
filter: ip and ( port 5060 )
#
U 10.1.99.153:5060 -> 10.1.99.154:5060
INVITE sip:6564564564 at 10.1.99.154:5060;user=phone SIP/2.0.
Record-Route: <sip:10.1.99.153;ftag=17188800-8754E9FD;lr=on>.
Via: SIP/2.0/UDP 10.1.99.153;branch=z9hG4bK6bd3.9c101652.0.
Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKfa0777c3A3756946.
From: "555123456" <sip:555123456 at 10.1.99.153>;tag=17188800-8754E9FD.
To: <sip:6564564564 at 10.1.99.153;user=phone>.
CSeq: 2 INVITE.
Call-ID: 220d1a74-64e4edba-a46ce337 at 10.1.99.161.
Contact: <sip:555123456 at 10.1.99.161:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.6.0036.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Max-Forwards: 69.
Content-Type: application/sdp.
Content-Length: 247.
.
v=0.
o=- 1156351584 1156351584 IN IP4 10.1.99.161.
s=Polycom IP Phone.
c=IN IP4 10.1.99.161.
t=0 0.
a=sendrecv.
m=audio 2260 RTP/AVP 0 8 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
#
U 10.1.99.154:5060 -> 10.1.99.153:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.1.99.153;branch=z9hG4bK6bd3.9c101652.0;received=10.1.99.153.
Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKfa0777c3A3756946.
From: "555123456" <sip:555123456 at 10.1.99.153>;tag=17188800-8754E9FD.
To: <sip:6564564564 at 10.1.99.153;user=phone>.
Call-ID: 220d1a74-64e4edba-a46ce337 at 10.1.99.161.
CSeq: 2 INVITE.
User-Agent: MediaGWTEST.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:6564564564 at 10.1.99.154>.
Content-Length: 0.
.
#
U 10.1.99.154:5060 -> 10.1.99.153:5060
SIP/2.0 603 Declined.
Via: SIP/2.0/UDP 10.1.99.153;branch=z9hG4bK6bd3.9c101652.0;received=10.1.99.153.
Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKfa0777c3A3756946.
From: "555123456" <sip:555123456 at 10.1.99.153>;tag=17188800-8754E9FD.
To: <sip:6564564564 at 10.1.99.153;user=phone>;tag=as3b8dae9d.
Call-ID: 220d1a74-64e4edba-a46ce337 at 10.1.99.161.
CSeq: 2 INVITE.
User-Agent: MediaGWTEST.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:6564564564 at 10.1.99.154>.
Content-Length: 0.
.
#
U 10.1.99.153:5060 -> 10.1.99.154:5060
ACK sip:6564564564 at 10.1.99.154:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.1.99.153;branch=z9hG4bK6bd3.9c101652.0.
From: "555123456" <sip:555123456 at 10.1.99.153>;tag=17188800-8754E9FD.
Call-ID: 220d1a74-64e4edba-a46ce337 at 10.1.99.161.
To: <sip:6564564564 at 10.1.99.153;user=phone>;tag=as3b8dae9d.
CSeq: 2 ACK.
User-Agent: OpenSer (1.0.1-tls (i386/linux)).
Content-Length: 0.
.
exit
8 received, 0 dropped
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