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Dear all,<br>
<br>
I have a very strange problem :<br>
<ul>
<li>external calls work perfectly,</li>
<li>internal calls between some phones too,</li>
<li>but internal call between two similar phones don't work !!!
(Snom 710)<br>
</li>
</ul>
When we have sound, there are no errors in asterisk. When we do not
have sound, there is the following error :<br>
<ul>
<li>[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.</li>
</ul>
This is a working internal call :<br>
<blockquote type="cite"> == Using SIP RTP CoS mark 5<br>
-- Executing [301@local:1] Dial("SIP/dbucher-00000000",
"SIP/phone1") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called phone1<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 answered SIP/dbucher-00000000<br>
-- Remotely bridging SIP/dbucher-00000000 and
SIP/phone1-00000001<br>
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)<br>
Got RTP packet from 192.168.128.99:49646 (type 126, seq
031575, ts 000001, len 000000)<br>
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190
ast_rtp_read: Unknown RTP codec 126 received from
'192.168.128.99:49646'<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)<br>
== Spawn extension (local, 301, 1) exited non-zero on
'SIP/dbucher-00000000'</blockquote>
This is a non-working call :<br>
<blockquote type="cite"> == Using SIP RTP CoS mark 5<br>
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No
SRTP module loaded, can't setup SRTP session.<br>
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-00000002",
"SIP/phone1") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called phone1<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 answered SIP/hsolutionspf5-00000002<br>
-- Remotely bridging SIP/hsolutionspf5-00000002 and
SIP/phone1-00000003<br>
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 000033)<br>
== Spawn extension (local, 301, 1) exited non-zero on
'SIP/hsolutionspf5-00000002'</blockquote>
I tried many options to disable SRTP but without success :<br>
<ul>
<li>canreinvite = no</li>
<li> canreinvite = nonat</li>
<li> srtpcapable=no</li>
<li>encryption=no</li>
<li>directmedia=nonat</li>
<li>...or noload => res_srtp.so in modules.conf<br>
</li>
</ul>
<br>
Any help would be GREATLY appreciated !<br>
<br>
Denis<br>
<br>
P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)<br>
<br>
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