Raspbx -> SPA112 -> SIP PHONES

neuberfran · August 30, 2016, 3:43pm

I staty try run RASPBX in RPi 2 B v1.1 ! with asterisk 11 and I have issues:
https://drive.google.com/file/d/0B9gPdWq4OH-XWXdhcjVmSmhYTnc/view?usp=sharing
In debian with wheezy image I install asterisk 1.8 with make/make install and everything worked
I put sip.conf/extensions.conf and rtp.conf from RPi with wheezy image to
RASPBX. I prefer cli commands and not graphics commands
https://sourceforge.net/p/raspbx/discussion/general/thread/1c13370b/1
Help me please

johnkiniston · August 30, 2016, 4:39pm

What is your question?

neuberfran · August 31, 2016, 3:06am

SRTP - Can’t provide secure audio requested in SDP offer

See image of the google drive please

jcolp · August 31, 2016, 10:02am

That means Asterisk has been configured for SRTP but the device in question did not offer it. What is the output of “sip set debug on” when you make a call attempt as well as the configuration in Asterisk for the device?

neuberfran · August 31, 2016, 4:25pm

[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] netsock2.c: == Using SIP RTP CoS mark 5
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 9
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 0
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 8
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 18
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 102
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 3
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found RTP audio format 101
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format G722 for ID 9
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format PCMU for ID 0
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format PCMA for ID 8
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format G729 for ID 18
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format ILBC for ID 102
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format GSM for ID 3
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Found audio description format telephone-event for ID 101
[2016-08-31 17:05:32] WARNING[1970][C-00000005] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjItX8yNwHvSeTHu3VL2RBcluoP-L5ftpz;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=kHxH039YCgG4lcH33QVurqEuAKRq3bIW
To: sip:1002@192.168.1.115;tag=as02c8a514
Call-ID: VbKjXPn92NfkwIWkt4jgqU6NKOURqOnF
CSeq: 16802 INVITE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2016-08-31 17:05:32] VERBOSE[1970][C-00000005] chan_sip.c: Scheduling destruction of SIP dialog ‘VbKjXPn92NfkwIWkt4jgqU6NKOURqOnF’ in 32000 ms (Method: INVITE)
[2016-08-31 17:05:32] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
ACK sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjItX8yNwHvSeTHu3VL2RBcluoP-L5ftpz
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=kHxH039YCgG4lcH33QVurqEuAKRq3bIW
To: sip:1002@192.168.1.115;tag=as02c8a514
Call-ID: VbKjXPn92NfkwIWkt4jgqU6NKOURqOnF
CSeq: 16802 ACK
Route: sip:192.168.1.115;lr
Content-Length: 0

<------------->
[2016-08-31 17:05:32] VERBOSE[1970] chan_sip.c: — (9 headers 0 lines) —
[2016-08-31 17:05:47] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>

<------------->
[2016-08-31 17:06:04] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘VbKjXPn92NfkwIWkt4jgqU6NKOURqOnF’ Method: ACK
[2016-08-31 17:07:08] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>

<------------->
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjBOOOnS0H4bB8fNmq6a8Hpr8rWZfFtwLr
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=CnKul2N7YvZFUqzr.R5dUupzbzmFf74-
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB
CSeq: 9508 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Content-Length: 0

<------------->
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: — (16 headers 0 lines) —
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: list_route: hop: sip:1002@192.168.1.100:5063;ob
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjBOOOnS0H4bB8fNmq6a8Hpr8rWZfFtwLr;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=CnKul2N7YvZFUqzr.R5dUupzbzmFf74-
To: sip:1002@192.168.1.115;tag=as27e0d6e5
Call-ID: Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB
CSeq: 9508 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4fc8bc1c"
Content-Length: 0

<------------>
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Scheduling destruction of SIP dialog ‘Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB’ in 32000 ms (Method: SUBSCRIBE)
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjK7ama2pOsrXvb93VdDseQFC6b.ox9nDK
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=CnKul2N7YvZFUqzr.R5dUupzbzmFf74-
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB
CSeq: 9509 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“4fc8bc1c”, uri="sip:1002@192.168.1.115", response=“0ef01c6ec83a2db22ea2f24171d0056b”, algorithm=MD5
Content-Length: 0

<------------->
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: — (17 headers 0 lines) —
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjK7ama2pOsrXvb93VdDseQFC6b.ox9nDK;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=CnKul2N7YvZFUqzr.R5dUupzbzmFf74-
To: sip:1002@192.168.1.115;tag=as27e0d6e5
Call-ID: Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB
CSeq: 9509 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2016-08-31 17:08:11] NOTICE[1970] chan_sip.c: Received SIP subscribe for peer without mailbox: 1002
[2016-08-31 17:08:11] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘Ej5sCM.VwO-3eQAR-XKR.0yx.3oqCEHB’ Method: SUBSCRIBE
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
REGISTER sip:192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjstvacBWX9cdVBXc51VsUnKtcDPbUe2zU
Route: sip:192.168.1.115;lr
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=EBZ06z.rcCw36HFCYr12ZyVsvt9-Zqj4
To: sip:1002@192.168.1.115
Call-ID: PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33
CSeq: 46542 REGISTER
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“3e089902”, uri=“sip:192.168.1.115”, response=“c5e8bffcd0c638be53cf21ebb0389fb9”, algorithm=MD5
User-Agent: CSipSimple_klte-23/r2457
Contact: sip:1002@192.168.1.100:5063;ob
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: — (14 headers 0 lines) —
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjstvacBWX9cdVBXc51VsUnKtcDPbUe2zU;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=EBZ06z.rcCw36HFCYr12ZyVsvt9-Zqj4
To: sip:1002@192.168.1.115;tag=as64a2fe45
Call-ID: PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33
CSeq: 46542 REGISTER
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="49d1f21b"
Content-Length: 0

<------------>
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Scheduling destruction of SIP dialog ‘PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33’ in 32000 ms (Method: REGISTER)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
REGISTER sip:192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjsxRraDz7Cj00t23wv-VlyQMHtSLNS9tY
Route: sip:192.168.1.115;lr
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=EBZ06z.rcCw36HFCYr12ZyVsvt9-Zqj4
To: sip:1002@192.168.1.115
Call-ID: PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33
CSeq: 46543 REGISTER
User-Agent: CSipSimple_klte-23/r2457
Contact: sip:1002@192.168.1.100:5063;ob
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“49d1f21b”, uri=“sip:192.168.1.115”, response=“57ab15fd15c43fe6eba0b163d7b383cf”, algorithm=MD5
Content-Length: 0

<------------->
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: — (14 headers 0 lines) —
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjsxRraDz7Cj00t23wv-VlyQMHtSLNS9tY;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=EBZ06z.rcCw36HFCYr12ZyVsvt9-Zqj4
To: sip:1002@192.168.1.115;tag=as64a2fe45
Call-ID: PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33
CSeq: 46543 REGISTER
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 900
Contact: sip:1002@192.168.1.100:5063;ob;expires=900
Date: Wed, 31 Aug 2016 16:08:12 GMT
Content-Length: 0

<------------>
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Scheduling destruction of SIP dialog ‘PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33’ in 32000 ms (Method: REGISTER)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjv9hd6iAOv2SePPXVkok80lcS21DUR2IZ
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=b620fTXcGVy00Mpr3u6uqEbk6HdWk6o.
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: xedFowdid7W3vlPsT5U8Uf19AtL1mTlv
CSeq: 20153 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Content-Length: 0

<------------->
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: — (16 headers 0 lines) —
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: list_route: hop: sip:1002@192.168.1.100:5063;ob
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjv9hd6iAOv2SePPXVkok80lcS21DUR2IZ;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=b620fTXcGVy00Mpr3u6uqEbk6HdWk6o.
To: sip:1002@192.168.1.115;tag=as2d7005da
Call-ID: xedFowdid7W3vlPsT5U8Uf19AtL1mTlv
CSeq: 20153 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="080b2293"
Content-Length: 0

<------------>
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Scheduling destruction of SIP dialog ‘xedFowdid7W3vlPsT5U8Uf19AtL1mTlv’ in 32000 ms (Method: SUBSCRIBE)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjXg.VdofQ28T6JIsKDCadmM4nve1Dywfz
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=b620fTXcGVy00Mpr3u6uqEbk6HdWk6o.
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: xedFowdid7W3vlPsT5U8Uf19AtL1mTlv
CSeq: 20154 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“080b2293”, uri="sip:1002@192.168.1.115", response=“bea0550753cd82461b6477c8e057b31b”, algorithm=MD5
Content-Length: 0

<------------->
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: — (17 headers 0 lines) —
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjXg.VdofQ28T6JIsKDCadmM4nve1Dywfz;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=b620fTXcGVy00Mpr3u6uqEbk6HdWk6o.
To: sip:1002@192.168.1.115;tag=as2d7005da
Call-ID: xedFowdid7W3vlPsT5U8Uf19AtL1mTlv
CSeq: 20154 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2016-08-31 17:08:12] NOTICE[1970] chan_sip.c: Received SIP subscribe for peer without mailbox: 1002
[2016-08-31 17:08:12] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘xedFowdid7W3vlPsT5U8Uf19AtL1mTlv’ Method: SUBSCRIBE
[2016-08-31 17:08:44] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘PwrIaV9MB4mf9-ysoC82LDcB3S5dSq33’ Method: REGISTER
[2016-08-31 17:12:02] VERBOSE[1935] asterisk.c: – Remote UNIX connection
[2016-08-31 17:12:02] VERBOSE[3783] asterisk.c: – Remote UNIX connection disconnected
[2016-08-31 17:12:02] VERBOSE[1935] asterisk.c: – Remote UNIX connection
[2016-08-31 17:12:02] VERBOSE[3786] asterisk.c: – Remote UNIX connection disconnected
[2016-08-31 17:12:31] VERBOSE[1935] asterisk.c: – Remote UNIX connection
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
INVITE sip:1004@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPja2ddp.rvBRMJRnlgDrlHPl.3QV-vK0sj
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13171 INVITE
Route: sip:192.168.1.115;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_klte-23/r2457
Content-Type: application/sdp
Content-Length: 601

v=0
o=- 3681648780 3681648780 IN IP4 192.168.1.100
s=pjmedia
c=IN IP4 192.168.1.100
t=0 0
m=audio 4012 RTP/AVP 9 0 8 18 102 3 101
c=IN IP4 192.168.1.100
a=rtcp:4013 IN IP4 192.168.1.100
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:imLTPriv3smujs8SePGw8HuLe2jirmy6W1M/9QZN
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:TjeZMXv7/ReiVD3HaPRQjC1reSqM1NAh89uSMpDW
<------------->
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c: — (16 headers 21 lines) —
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Using INVITE request as basis request - ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPja2ddp.rvBRMJRnlgDrlHPl.3QV-vK0sj;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115;tag=as5afcd5b5
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13171 INVITE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7bff622f"
Content-Length: 0

<------------>
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog ‘ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha’ in 32000 ms (Method: INVITE)
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
ACK sip:1004@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPja2ddp.rvBRMJRnlgDrlHPl.3QV-vK0sj
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115;tag=as5afcd5b5
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13171 ACK
Route: sip:192.168.1.115;lr
Content-Length: 0

<------------->
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c: — (9 headers 0 lines) —
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
INVITE sip:1004@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjIKOX.pav9JZBTIc8kfBnDqJekcBS6l21
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13172 INVITE
Route: sip:192.168.1.115;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_klte-23/r2457
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“7bff622f”, uri="sip:1004@192.168.1.115", response=“574705922494320189399f1c0b13c5dc”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 601

v=0
o=- 3681648780 3681648780 IN IP4 192.168.1.100
s=pjmedia
c=IN IP4 192.168.1.100
t=0 0
m=audio 4012 RTP/AVP 9 0 8 18 102 3 101
c=IN IP4 192.168.1.100
a=rtcp:4013 IN IP4 192.168.1.100
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:imLTPriv3smujs8SePGw8HuLe2jirmy6W1M/9QZN
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:TjeZMXv7/ReiVD3HaPRQjC1reSqM1NAh89uSMpDW
<------------->
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c: — (17 headers 21 lines) —
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Using INVITE request as basis request - ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] netsock2.c: == Using SIP RTP CoS mark 5
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 9
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 0
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 8
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 18
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 102
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 3
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found RTP audio format 101
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format G722 for ID 9
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format PCMU for ID 0
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format PCMA for ID 8
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format G729 for ID 18
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format ILBC for ID 102
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format GSM for ID 3
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Found audio description format telephone-event for ID 101
[2016-08-31 17:12:43] WARNING[1970][C-00000006] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c:
<— Reliably Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjIKOX.pav9JZBTIc8kfBnDqJekcBS6l21;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115;tag=as5afcd5b5
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13172 INVITE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2016-08-31 17:12:43] VERBOSE[1970][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog ‘ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha’ in 32000 ms (Method: INVITE)
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
ACK sip:1004@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjIKOX.pav9JZBTIc8kfBnDqJekcBS6l21
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=6N5ALasDQkcDb7VgSUkz.Dcv3E5pIuTf
To: sip:1004@192.168.1.115;tag=as5afcd5b5
Call-ID: ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha
CSeq: 13172 ACK
Route: sip:192.168.1.115;lr
Content-Length: 0

<------------->
[2016-08-31 17:12:43] VERBOSE[1970] chan_sip.c: — (9 headers 0 lines) —
[2016-08-31 17:12:49] VERBOSE[3853] asterisk.c: – Remote UNIX connection disconnected
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjI0hLc8PnpspRZIDb8pW1HmhCkTOMNNxA
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=yGahnw4FbJ.vyGa4aBhnOLX5KQThIKVO
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc
CSeq: 1696 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Content-Length: 0

<------------->
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: — (16 headers 0 lines) —
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: list_route: hop: sip:1002@192.168.1.100:5063;ob
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjI0hLc8PnpspRZIDb8pW1HmhCkTOMNNxA;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=yGahnw4FbJ.vyGa4aBhnOLX5KQThIKVO
To: sip:1002@192.168.1.115;tag=as19d74d8c
Call-ID: r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc
CSeq: 1696 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0f18bb5f"
Content-Length: 0

<------------>
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Scheduling destruction of SIP dialog ‘r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc’ in 32000 ms (Method: SUBSCRIBE)
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>
SUBSCRIBE sip:1002@192.168.1.115 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5063;rport;branch=z9hG4bKPjl8ec2nuuiCKNGT4LC-ysFy691cp6qNDH
Max-Forwards: 70
From: sip:1002@192.168.1.115;tag=yGahnw4FbJ.vyGa4aBhnOLX5KQThIKVO
To: sip:1002@192.168.1.115
Contact: sip:1002@192.168.1.100:5063;ob
Call-ID: r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc
CSeq: 1697 SUBSCRIBE
Route: sip:192.168.1.115;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_klte-23/r2457
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“0f18bb5f”, uri="sip:1002@192.168.1.115", response=“68ad615768283eb41123a483fea0fa3f”, algorithm=MD5
Content-Length: 0

<------------->
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: — (17 headers 0 lines) —
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Creating new subscription
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Sending to 192.168.1.100:5063 (NAT)
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Found peer ‘1002’ for ‘1002’ from 192.168.1.100:5063
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c:
<— Transmitting (NAT) to 192.168.1.100:5063 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.100:5063;branch=z9hG4bKPjl8ec2nuuiCKNGT4LC-ysFy691cp6qNDH;received=192.168.1.100;rport=5063
From: sip:1002@192.168.1.115;tag=yGahnw4FbJ.vyGa4aBhnOLX5KQThIKVO
To: sip:1002@192.168.1.115;tag=as19d74d8c
Call-ID: r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc
CSeq: 1697 SUBSCRIBE
Server: Asterisk PBX 11.21.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2016-08-31 17:13:07] NOTICE[1970] chan_sip.c: Received SIP subscribe for peer without mailbox: 1002
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘r3EhBH4A6JTmuYMhEqwpCPZgyqjg.ZIc’ Method: SUBSCRIBE
[2016-08-31 17:13:07] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>

<------------->
[2016-08-31 17:13:15] VERBOSE[1970] chan_sip.c: Really destroying SIP dialog ‘ZuvPR8voFlCnTlkL1Vj0MFpj-Ymj8fha’ Method: ACK
[2016-08-31 17:14:27] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>

<------------->
[2016-08-31 17:15:48] VERBOSE[1970] chan_sip.c:
<— SIP read from UDP:192.168.1.100:5063 —>

<------------->

jcolp · August 31, 2016, 4:43pm

The client has been configured with optional SRTP support which is not supported by chan_sip. You will need to disable it on the client, or enable mandatory SRTP and configure sip.conf accordingly if it is not yet already configured for it.

neuberfran · August 31, 2016, 6:54pm

Please
How do I do that?

jcolp · August 31, 2016, 6:56pm

I’ve never used the SIP client you are using so I do not know how to configure it or where the option would be.

neuberfran · August 31, 2016, 7:15pm

My client is RASPBX?

Will My problem can solve if I change this screen configurations?
https://drive.google.com/file/d/0B9gPdWq4OH-XN1U1MjVOdWJrZlU/view?usp=sharing

jcolp · August 31, 2016, 7:17pm

No, that appears to be FreePBX or something similar. The call in question is coming from something which identifies itself as “CSipSimple_klte-23/r2457”. That is what needs to be changed.

neuberfran · August 31, 2016, 7:45pm

Yes, I have CSipSimple app in my cell to make calls to my PC with ZOIPER (sipPHONE)

From ZOIPER to CSipSimple it’s OK. But from CSip to ZOIPER NO

jcolp · August 31, 2016, 7:46pm

The CSipSimple app is the problem. It needs to have encryption disabled.

neuberfran · September 1, 2016, 2:21am

Please,
how to I disable srtp module in freepbx(raspbx)

johnkiniston · September 1, 2016, 2:36am

If you are looking for FreePBX Support I’d recommend checking their forum at http://community.freepbx.org

satish4asterisk · September 1, 2016, 10:14am

Here is one link I found googling which you can use to disable srtp on csipsimple

disable srtp and zrtp in your client.

neuberfran · September 10, 2016, 5:03pm

SOLVED thanks guys for everything