Dear Sam, dear jg, dear Mitul, dear all,
Thanks a lot for your advices! I had the same idea, but it still doesn't
work!
Maybe I changed the wrong option on the GUI configuration ?
I went to menu "Setup" > "Identity 1" > "RTP" > "RTP Encryption:" >
"off" on both phones.
And in the configuration I see "user_srtp1!: off"
Is this right ?
Denis
Le 12.11.2015 17:05, Sam Basan a écrit :
Snom default configuration is SRTP enabled.
You should disable the SRTP from the phone web GUI configuration
**
**
*Sincerely,*
cid:[email protected]
*Sam Basan*
cid:[email protected]
*From:*Mitul Limbani [mailto:[email protected]]
*Sent:* Thursday, November 12, 2015 5:25 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
*Subject:* Re: [asterisk-users] No sound with internal calls depending
on which phones
You might have to disable srtp negotiations inside the phone web ui
options.
Mitul
On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER"
<[email protected] <mailto:[email protected]>> wrote:
Dear all,
I have a very strange problem :
* external calls work perfectly,
* internal calls between some phones too,
* but internal call between two similar phones don't work !!!
(Snom 710)
When we have sound, there are no errors in asterisk. When we do
not have sound, there is the following error :
* [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.
This is a working internal call :
== Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-00000000",
"SIP/phone1") in new stack
== Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-00000001 is ringing
-- SIP/phone1-00000001 is ringing
-- SIP/phone1-00000001 is ringing
-- SIP/phone1-00000001 is ringing
-- SIP/phone1-00000001 is ringing
-- SIP/phone1-00000001 answered SIP/dbucher-00000000
-- Remotely bridging SIP/dbucher-00000000 and
SIP/phone1-00000001
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646
<http://192.168.128.99:49646> (type 00, len 000160)
Got RTP packet from 192.168.128.99:49646
<http://192.168.128.99:49646> (type 126, seq 031575, ts
000001, len 000000)
[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 <http://192.168.128.99:49646>'
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818
<http://192.168.128.231:57818> (type 00, len 000160)
== Spawn extension (local, 301, 1) exited non-zero on
'SIP/dbucher-00000000'
This is a non-working call :
== Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp:
No SRTP module loaded, can't setup SRTP session.
-- Executing [301@local:1]
Dial("SIP/hsolutionspf5-00000002", "SIP/phone1") in new stack
== Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-00000003 is ringing
-- SIP/phone1-00000003 is ringing
-- SIP/phone1-00000003 is ringing
-- SIP/phone1-00000003 is ringing
-- SIP/phone1-00000003 is ringing
-- SIP/phone1-00000003 answered SIP/hsolutionspf5-00000002
-- Remotely bridging SIP/hsolutionspf5-00000002 and
SIP/phone1-00000003
Sent RTP P2P packet to 192.168.128.228:65494
<http://192.168.128.228:65494> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494
<http://192.168.128.228:65494> (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
Sent RTP P2P packet to 192.168.128.231:51350
<http://192.168.128.231:51350> (type 03, len 000033)
== Spawn extension (local, 301, 1) exited non-zero on
'SIP/hsolutionspf5-00000002'
I tried many options to disable SRTP but without success :
* canreinvite = no
* canreinvite = nonat
* srtpcapable=no
* encryption=no
* directmedia=nonat
* ...or noload => res_srtp.so in modules.conf
Any help would be GREATLY appreciated !
Denis
P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)
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