Do not start rtp from qutecom -> asterisk

Hello!

before we used asterisk 1.8 and it worked fine.

but now i installed asterisk pbx 11 /debian/

my scheme:
Asterisk (pbx 11) --LAN(10.16.66.0/255)–sip clients (softphones: qutecom, microsip)
iptables is empty, full accept

when i use qutecom for outgoing calls to another sip/client or outside - no have sound!
i do not see rtp traffic from qutecom to asterisk (nothing on a both machines (qc) and (aster)), but we have traff from asterisk to qutecom.

but when i use microsip or jitsi or outside incoming for qurecom calls - rtp session is start ! and all is fine.

Calling from qutecom to jitsi console log

asterisk*CLI> sip set debug ip 10.16.66.33
SIP Debugging Enabled for IP: 10.16.66.33

<— SIP read from UDP:10.16.66.33:19459 —>
OPTIONS sip:10.16.66.10 SIP/2.0
Call-ID: f6eb4dea06729f485c7a2ede824a4940@0:0:0:0:0:0:0:0
CSeq: 642 OPTIONS
From: “100” sip:100@10.16.66.10;tag=264b1af9
To: “100” sip:100@10.16.66.10
Via: SIP/2.0/UDP 10.16.66.33:19459;branch=z9hG4bK-353834-65d60e9e207e05fcbec3a2b515aae5dc
Max-Forwards: 70
Contact: “100” sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
User-Agent: Jitsi1.0-beta1-nightly.build.3820Linux
Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Looking for s in IN (domain 10.16.66.10)

<— Transmitting (no NAT) to 10.16.66.33:19459 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.16.66.33:19459;branch=z9hG4bK-353834-65d60e9e207e05fcbec3a2b515aae5dc;received=10.16.66.33
From: “100” sip:100@10.16.66.10;tag=264b1af9
To: “100” sip:100@10.16.66.10;tag=as181af0cf
Call-ID: f6eb4dea06729f485c7a2ede824a4940@0:0:0:0:0:0:0:0
CSeq: 642 OPTIONS
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘f6eb4dea06729f485c7a2ede824a4940@0:0:0:0:0:0:0:0’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘42c87bf53a4a25292f887bb13db6d4c8@10.16.66.10:5060’ Method: BYE
Audio is at 10042
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.16.66.33:19459:
INVITE sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK4d52fcd0
Max-Forwards: 70
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
Contact: sip:101@10.16.66.10:5060
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
CSeq: 102 INVITE
User-Agent: int9@r66.ru
Date: Mon, 11 Mar 2013 09:54:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 409

v=0
o=root 2138056753 2138056753 IN IP4 10.16.66.10
s=Asterisk PBX 11.2.1
c=IN IP4 10.16.66.10
t=0 0
m=audio 10042 RTP/AVP 0 18 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #1 (no NAT) to 10.16.66.33:19459:
INVITE sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK4d52fcd0
Max-Forwards: 70
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
Contact: sip:101@10.16.66.10:5060
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
CSeq: 102 INVITE
User-Agent: int9@r66.ru
Date: Mon, 11 Mar 2013 09:54:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 409

v=0
o=root 2138056753 2138056753 IN IP4 10.16.66.10
s=Asterisk PBX 11.2.1
c=IN IP4 10.16.66.10
t=0 0
m=audio 10042 RTP/AVP 0 18 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read from UDP:10.16.66.33:19459 —>
SIP/2.0 180 Ringing
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK4d52fcd0
CSeq: 102 INVITE
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
Contact: “100” sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
User-Agent: Jitsi1.0-beta1-nightly.build.3820Linux
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10

<— SIP read from UDP:10.16.66.33:19459 —>
SIP/2.0 180 Ringing
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK4d52fcd0
CSeq: 102 INVITE
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
Contact: “100” sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
User-Agent: Jitsi1.0-beta1-nightly.build.3820Linux
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10

<— SIP read from UDP:10.16.66.33:19459 —>
SIP/2.0 200 OK
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK4d52fcd0
CSeq: 102 INVITE
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
Contact: “100” sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
User-Agent: Jitsi1.0-beta1-nightly.build.3820Linux
Content-Type: application/sdp
Content-Length: 197

v=0
o=100 0 0 IN IP4 10.16.66.33
s=-
c=IN IP4 10.16.66.33
t=0 0
m=audio 5122 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=zrtp-hash:1.10 ca859041c47bda4bc576b6d6f8f9901ce18e8114c247b14dce3e1e2852f5a268
<------------->
— (10 headers 8 lines) —
Found RTP audio format 3
Found audio description format GSM for ID 3
Capabilities: us - (gsm|ulaw|alaw|g726|g729|g722), peer - audio=(gsm)/video=(nothing)/text=(nothing), combined - (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.16.66.33:5122
list_route: hop: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
set_destination: Parsing sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 for address/port to send to
set_destination: set destination to 10.16.66.33:19459
Transmitting (no NAT) to 10.16.66.33:19459:
ACK sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK6c1d367d
Max-Forwards: 70
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Contact: sip:101@10.16.66.10:5060
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
CSeq: 102 ACK
User-Agent: int9@r66.ru
Content-Length: 0


Really destroying SIP dialog ‘079953c3d1af90a52d8f1ea12dd53d9b@0:0:0:0:0:0:0:0’ Method: OPTIONS
Scheduling destruction of SIP dialog ‘0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 for address/port to send to
set_destination: set destination to 10.16.66.33:19459
Reliably Transmitting (no NAT) to 10.16.66.33:19459:
BYE sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK15482925
Max-Forwards: 70
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
CSeq: 103 BYE
User-Agent: int9@r66.ru
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:10.16.66.33:19459 —>
SIP/2.0 200 OK
To: sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10;tag=a461d359
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK15482925
CSeq: 103 BYE
Call-ID: 0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060
From: “Ruden” sip:101@10.16.66.10;tag=as54a159cc
Contact: “100” sip:100@10.16.66.33:19459;transport=udp;registering_acc=10_16_66_10
User-Agent: Jitsi1.0-beta1-nightly.build.3820Linux
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘0dc0c523499c90d0633ded1f3eacad1c@10.16.66.10:5060’ Method: INVITE
asterisk*CLI> sip set debug off
SIP Debugging Disabled

Why is qutecom rtp not work now ?

There are no media related problems that I can see in that trace. There is a retransmission, which means either your network is overloaded or the phone is not sending Trying properly.

thanks for answer

this is no network, before all worked fine (several years), and with other softphones are working well now

and this is now qutecom , becouse i have 5 stations with same soft and same problems

when im using “tcpdump -T rtp” on qutecom’s machine i dont see outgoing traf, only income from asterisk.

maybe qutecom is waiting some data (act, invite) from asterisk for start session ?

That trace only shows a JITSI user agent and that has reponded to the INVITE from Asterisk, telling it to send RTP, in GSM format, to port 1042 at 10.16.66.10.

c=IN IP4 10.16.66.10
m=audio 10042 RTP/AVP 0 18 8 3 111 9 101

this mean that qutecom do not ready , and i think this is becouse asterisk do not correct INVITE qutecom for ready

what can i do for fix this problem?

Im sory

.33 (before) console data for JItsi machine

now i posting .35 data qutecom’s machine

SIP Debugging Enabled for IP: 10.16.66.35

<— SIP read from UDP:10.16.66.35:5060 —>
INVITE sip:10@10.16.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK675821535
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10
Call-ID: 1760756938@10.16.66.35
CSeq: 20 INVITE
Contact: sip:101@10.16.66.35:5060
Max-Forwards: 70
User-Agent: qutecom/revg/trunk/
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 368

v=0
o=userX 20000001 20000001 IN IP4 10.16.66.35
s=A call
c=IN IP4 10.16.66.35
t=1363002303 1363005903
m=audio 10600 RTP/AVP 0 8 3 9 101
a=ptime:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
m=video 10702 RTP/AVP 34 31
a=rtpmap:34 H263/90000/1
a=rtpmap:31 H261/90000/1
<------------->
— (13 headers 15 lines) —
Sending to 10.16.66.35:5060 (no NAT)
Using INVITE request as basis request - 1760756938@10.16.66.35
Found peer ‘101’ for ‘101’ from 10.16.66.35:5060

<— Reliably Transmitting (no NAT) to 10.16.66.35:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.16.66.35:5060;branch=z9hG4bK675821535;received=10.16.66.35;rport=5060
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10;tag=as4f699053
Call-ID: 1760756938@10.16.66.35
CSeq: 20 INVITE
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="28f32888"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1760756938@10.16.66.35’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.16.66.35:5060 —>
ACK sip:10@10.16.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK675821535
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10;tag=as4f699053
Call-ID: 1760756938@10.16.66.35
CSeq: 20 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:10.16.66.35:5060 —>
INVITE sip:10@10.16.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK1028272922
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10
Call-ID: 1760756938@10.16.66.35
CSeq: 21 INVITE
Contact: sip:101@10.16.66.35:5060
Authorization: Digest username=“101”, realm=“asterisk”, nonce=“28f32888”, uri="sip:10@10.16.66.10", response=“2c81e4cd9240e35de937898ddf60da25”, algorithm=MD5
Max-Forwards: 70
User-Agent: qutecom/revg/trunk/
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 368

v=0
o=userX 20000001 20000001 IN IP4 10.16.66.35
s=A call
c=IN IP4 10.16.66.35
t=1363002303 1363005903
m=audio 10600 RTP/AVP 0 8 3 9 101
a=ptime:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
m=video 10702 RTP/AVP 34 31
a=rtpmap:34 H263/90000/1
a=rtpmap:31 H261/90000/1
<------------->
— (14 headers 15 lines) —
Sending to 10.16.66.35:5060 (no NAT)
Using INVITE request as basis request - 1760756938@10.16.66.35
Found peer ‘101’ for ‘101’ from 10.16.66.35:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found RTP video format 31
Found video description format H263 for ID 34
Found video description format H261 for ID 31
Capabilities: us - (gsm|ulaw|alaw|g726|g729|g722), peer - audio=(gsm|ulaw|alaw|g722)/video=(h261|h263)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.16.66.35:10600
Looking for 10 in OUT (domain 10.16.66.10)
list_route: hop: sip:101@10.16.66.35:5060

<— Transmitting (no NAT) to 10.16.66.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.66.35:5060;branch=z9hG4bK1028272922;received=10.16.66.35;rport=5060
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10
Call-ID: 1760756938@10.16.66.35
CSeq: 21 INVITE
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10@10.16.66.10:5060
Content-Length: 0

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

<— Transmitting (no NAT) to 10.16.66.35:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.16.66.35:5060;branch=z9hG4bK1028272922;received=10.16.66.35;rport=5060
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10;tag=as6ff92f33
Call-ID: 1760756938@10.16.66.35
CSeq: 21 INVITE
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10@10.16.66.10:5060
Content-Length: 0

<------------>
Audio is at 10096
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100012 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.35:5060;branch=z9hG4bK1028272922;received=10.16.66.35;rport=5060
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10;tag=as6ff92f33
Call-ID: 1760756938@10.16.66.35
CSeq: 21 INVITE
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10@10.16.66.10:5060
Content-Type: application/sdp
Content-Length: 352

v=0
o=root 946271702 946271702 IN IP4 10.16.66.10
s=Asterisk PBX 11.2.1
c=IN IP4 10.16.66.10
t=0 0
m=audio 10096 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34 31
<------------>

<— SIP read from UDP:10.16.66.35:5060 —>
ACK sip:10@10.16.66.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK2006811596
From: Ruden sip:101@10.16.66.10;tag=1199711224
To: sip:10@10.16.66.10;tag=as6ff92f33
Call-ID: 1760756938@10.16.66.35
CSeq: 21 ACK
Contact: sip:101@10.16.66.35:5060
Max-Forwards: 70
User-Agent: qutecom/revg/trunk/
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Reliably Transmitting (no NAT) to 10.16.66.35:5060:
OPTIONS sip:101@10.16.66.35:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK58ea80de
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.16.66.10;tag=as30995ccf
To: sip:101@10.16.66.35:5060
Contact: sip:asterisk@10.16.66.10:5060
Call-ID: 21a00da439c10b23468965ba74a0708f@10.16.66.10:5060
CSeq: 102 OPTIONS
User-Agent: int9@r66.ru
Date: Mon, 11 Mar 2013 11:44:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK58ea80de
From: “asterisk” sip:asterisk@10.16.66.10;tag=as30995ccf
To: sip:101@10.16.66.35:5060
Call-ID: 21a00da439c10b23468965ba74a0708f@10.16.66.10:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK58ea80de
From: “asterisk” sip:asterisk@10.16.66.10;tag=as30995ccf
To: sip:101@10.16.66.35:5060;tag=1631545789
Call-ID: 21a00da439c10b23468965ba74a0708f@10.16.66.10:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘21a00da439c10b23468965ba74a0708f@10.16.66.10:5060’ Method: OPTIONS
Scheduling destruction of SIP dialog ‘1760756938@10.16.66.35’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:101@10.16.66.35:5060 for address/port to send to
set_destination: set destination to 10.16.66.35:5060
Reliably Transmitting (no NAT) to 10.16.66.35:5060:
BYE sip:101@10.16.66.35:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK220895d4;rport
Max-Forwards: 70
From: sip:10@10.16.66.10;tag=as6ff92f33
To: Ruden sip:101@10.16.66.10;tag=1199711224
Call-ID: 1760756938@10.16.66.35
CSeq: 102 BYE
User-Agent: int9@r66.ru
Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.16.66.10”, nonce=“28f32888”, response="2500e6c5e4c19341f2eed4b3581868e8"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK220895d4;rport=5060
From: sip:10@10.16.66.10;tag=as6ff92f33
To: Ruden sip:101@10.16.66.10;tag=1199711224
Call-ID: 1760756938@10.16.66.35
CSeq: 102 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK220895d4;rport=5060
From: sip:10@10.16.66.10;tag=as6ff92f33
To: Ruden sip:101@10.16.66.10;tag=1199711224
Call-ID: 1760756938@10.16.66.35
CSeq: 102 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘1760756938@10.16.66.35’ Method: ACK

<— SIP read from UDP:10.16.66.35:5060 —>
OPTIONS sip:101@10.16.66.10 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK1504634406
From: Ruden sip:101@10.16.66.10;tag=2085185077
To: sip:101@10.16.66.10
Call-ID: 1312989831@10.16.66.35
CSeq: 20 OPTIONS
Max-Forwards: 70
User-Agent: qutecom/revg/trunk/
Expires: 120
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Looking for 101 in IN (domain 10.16.66.10)

<— Transmitting (no NAT) to 10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK1504634406;received=10.16.66.35
From: Ruden sip:101@10.16.66.10;tag=2085185077
To: sip:101@10.16.66.10;tag=as62ce4b9c
Call-ID: 1312989831@10.16.66.35
CSeq: 20 OPTIONS
Server: int9@r66.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.16.66.10:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1312989831@10.16.66.35’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘938284602@10.16.66.35’ Method: OPTIONS
asterisk*CLI> sip set debug off
SIP Debugging Disabled

This confirms that the quetcom has accepted the SDP from Asterisk and therefore knows where to send to:

<--- SIP read from UDP:10.16.66.35:5060 ---> ACK sip:10@10.16.66.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.66.35:5060;rport;branch=z9hG4bK2006811596 From: Ruden <sip:101@10.16.66.10>;tag=1199711224 To: <sip:10@10.16.66.10>;tag=as6ff92f33 Call-ID: 1760756938@10.16.66.35 CSeq: 21 ACK Contact: <sip:101@10.16.66.35:5060> Max-Forwards: 70 User-Agent: qutecom/revg/trunk/ Content-Length: 0

Asterisk has told it to send to 10.16.66.10:10096

c=IN IP4 10.16.66.10
m=audio 10096 RTP/AVP 0 8 3 9 101

No ideas ?

c=IN IP4 10.16.66.10
t=0 0

is "t=0 0"ok?

this log when ( * =>> qutecom )
works’s fine

root@asterisk:~# rasterisk
Asterisk 11.2.1, Copyright © 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.2.1 currently running on asterisk (pid = 14533)
asterisk*CLI> sip set debug ip 10.16.66.35
SIP Debugging Enabled for IP: 10.16.66.35
Audio is at 10050
Adding codec 100008 (g729) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100011 (g726) to SDP
Adding codec 100012 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.16.66.35:5060:
INVITE sip:101@10.16.66.35:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK0c81bb80
Max-Forwards: 70
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060
Contact: sip:100@10.16.66.10:5060
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 102 INVITE
User-Agent: int9@r66.ru
Date: Mon, 11 Mar 2013 13:56:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 407

v=0
o=root 476868804 476868804 IN IP4 10.16.66.10
s=Asterisk PBX 11.2.1
c=IN IP4 10.16.66.10
t=0 0
m=audio 10050 RTP/AVP 18 0 8 3 111 9 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK0c81bb80
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK0c81bb80
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 102 INVITE
Contact: sip:101@10.16.66.35:5060
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:101@10.16.66.35:5060

<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK0c81bb80
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 102 INVITE
Contact: sip:101@10.16.66.35:5060
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Type: application/sdp
Content-Length: 282

v=0
o=userX 20000001 20000001 IN IP4 10.16.66.35
s=Asterisk PBX 11.2.1
c=IN IP4 10.16.66.35
t=0 0
m=audio 10600 RTP/AVP 0 8 3 9 101
a=ptime:20
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
<------------->
— (10 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726|g729|g722), peer - audio=(gsm|ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.16.66.35:10600
list_route: hop: sip:101@10.16.66.35:5060
set_destination: Parsing sip:101@10.16.66.35:5060 for address/port to send to
set_destination: set destination to 10.16.66.35:5060
Transmitting (no NAT) to 10.16.66.35:5060:
ACK sip:101@10.16.66.35:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK716a1b41
Max-Forwards: 70
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Contact: sip:100@10.16.66.10:5060
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 102 ACK
User-Agent: int9@r66.ru
Content-Length: 0


Scheduling destruction of SIP dialog ‘76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:101@10.16.66.35:5060 for address/port to send to
set_destination: set destination to 10.16.66.35:5060
Reliably Transmitting (no NAT) to 10.16.66.35:5060:
BYE sip:101@10.16.66.35:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK55b43e37
Max-Forwards: 70
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 103 BYE
User-Agent: int9@r66.ru
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK55b43e37
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 103 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.16.66.35:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.16.66.10:5060;branch=z9hG4bK55b43e37
From: sip:100@10.16.66.10;tag=as5f5bb9f2
To: sip:101@10.16.66.35:5060;tag=411943490
Call-ID: 76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060
CSeq: 103 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘76cd1a865f78989d64bd253e1a67aa8a@10.16.66.10:5060’ Method: INVITE
[Mar 11 19:56:56] NOTICE[14569]: chan_sip.c:14907 sip_reregister: – Re-registration for 99051000118794@sip.skype.com
[Mar 11 19:56:56] NOTICE[14569]: chan_sip.c:23198 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 120 sec (Scheduling reregistration in 105 s)
asteriskCLI> sip set debug off
SIP Debugging Disabled
asterisk
CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups