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Matthew Boehm wrote:
<blockquote cite="midBF2388F9.528E%25mboehm@cytelcom.com" type="cite">
<pre wrap="">Well, for one, what you describe is NOT a bug in RealTime SIP. (damn I wish
people would stop doing that.)
Secondly, this does not belong on the developers list.
Since your UA has obviously registered, SIP RT has done its job.
Your problem now seems to be in your extensions. Do you have an extensions
line for 18005551212?
</pre>
</blockquote>
Yes, I have a exten => _X. in the context that the sip device has as
context= in the sipusers table. However, the sipusers entry didn't get
loaded and the device tried to use default.<br>
<br>
default just hangs up on all calls sent to it.<br>
<br>
So, IMHO, SIP RT hasn't done it's job when a device with entries in
both the sippeers and sipusers families registers, and the sipusers
family entry doesn't get loaded.<br>
<br>
As for an update. I restarted asterisk and everything seems to be
working fine. I have yet to be able to reproduce this bug. However, I
won't really be testing it until sunday, so I will update this thread
sunday evening after we do extensive tests.<br>
<br>
-Chris<br>
<br>
<blockquote cite="midBF2388F9.528E%25mboehm@cytelcom.com" type="cite">
<pre wrap="">-Matthew
</pre>
<blockquote type="cite">
<pre wrap="">From: "Chris A. Icide" <a class="moz-txt-link-rfc2396E" href="mailto:chris@netgeeks.net"><chris@netgeeks.net></a>
Reply-To: Asterisk Developers Mailing List <a class="moz-txt-link-rfc2396E" href="mailto:asterisk-dev@lists.digium.com"><asterisk-dev@lists.digium.com></a>
Date: Fri, 12 Aug 2005 20:49:26 -0700
To: Asterisk Developers Mailing List <a class="moz-txt-link-rfc2396E" href="mailto:asterisk-dev@lists.digium.com"><asterisk-dev@lists.digium.com></a>
Subject: [Asterisk-Dev] Bug in realtime SIP
Asterisk version: CVS Head from 8/11/05
extconfig.conf
sipusers => mysql,zbx,sipusers
sippeers => mysql,zbx,sippeers
localhost*CLI> realtime mysql status
Connected to zbx@localhost, port 3306 with username zbx for 27 minutes,
20 seconds.
realtime load sippeers name 104 and realtime load sipusers 104 both show
the sip device information on screen.
When the sip device registers (it's dynamic), sip show peers and sip
show users show the following
sip show peers
104/104 (Unspecified) D 255.255.255.255
0 Unmonitored
sip show users
No entry every shows up for users. If I try to make a call with the
device, I get 403 Forbidden
<-- SIP read from 192.168.254.16:5060:
INVITE <a class="moz-txt-link-freetext" href="sip:18005551212@192.168.254.6">sip:18005551212@192.168.254.6</a> SIP/2.0
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <a class="moz-txt-link-rfc2396E" href="sip:104@192.168.254.6"><sip:104@192.168.254.6></a>;tag=ffbcad7047bef41co1
To: <a class="moz-txt-link-rfc2396E" href="sip:18005551212@192.168.254.6"><sip:18005551212@192.168.254.6></a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:11739700-af5297cc@192.168.254.16">11739700-af5297cc@192.168.254.16</a>
CSeq: 101 INVITE
Max-Forwards: 70
Contact: Sipura2000-2 <a class="moz-txt-link-rfc2396E" href="sip:104@192.168.254.16:5060"><sip:104@192.168.254.16:5060></a>
Expires: 240
User-Agent: Sipura/SPA2000-2.0.10(c)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 266317 266317 IN IP4 192.168.254.16
s=-
c=IN IP4 192.168.254.16
t=0 0
m=audio 16384 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (14 headers 19 lines)---
Using INVITE request as basis request - <a class="moz-txt-link-abbreviated" href="mailto:11739700-af5297cc@192.168.254.16">11739700-af5297cc@192.168.254.16</a>
Sending to 192.168.254.16 : 5060 (non-NAT)
Found no matching peer or user for '192.168.254.16:5060'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.254.16:16384
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 18005551212 in default
list_route: hop: <a class="moz-txt-link-rfc2396E" href="sip:104@192.168.254.16:5060"><sip:104@192.168.254.16:5060></a>
Transmitting (no NAT) to 192.168.254.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <a class="moz-txt-link-rfc2396E" href="sip:104@192.168.254.6"><sip:104@192.168.254.6></a>;tag=ffbcad7047bef41co1
To: <a class="moz-txt-link-rfc2396E" href="sip:18005551212@192.168.254.6"><sip:18005551212@192.168.254.6></a>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:11739700-af5297cc@192.168.254.16">11739700-af5297cc@192.168.254.16</a>
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <a class="moz-txt-link-rfc2396E" href="sip:18005551212@192.168.254.4"><sip:18005551212@192.168.254.4></a>
Content-Length: 0
---
-- Executing Hangup("SIP/192.168.254.6-08a17090", "") in new stack
== Spawn extension (default, 18005551212, 1) exited non-zero on
'SIP/192.168.254.6-08a17090'
Reliably Transmitting (no NAT) to 192.168.254.16:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.254.16:5060;branch=z9hG4bK-ba65de3c
From: Sipura2000-2 <a class="moz-txt-link-rfc2396E" href="sip:104@192.168.254.6"><sip:104@192.168.254.6></a>;tag=ffbcad7047bef41co1
To: <a class="moz-txt-link-rfc2396E" href="sip:18005551212@192.168.254.6"><sip:18005551212@192.168.254.6></a>;tag=as31d7c52b
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:11739700-af5297cc@192.168.254.16">11739700-af5297cc@192.168.254.16</a>
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <a class="moz-txt-link-rfc2396E" href="sip:18005551212@192.168.254.4"><sip:18005551212@192.168.254.4></a>
Content-Length: 0
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</pre>
</blockquote>
<pre wrap=""><!---->
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</pre>
</blockquote>
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