Hi, I am using asterisk version 11.5.1
Everything is working fine on my asterisk box.
When I set ANY dns(A good or a bad) in /etc/network/interfaces
auto eth0
iface eth0 inet static
address 10.0..
netmask 255.255.255.0
gateway 10.0..
dns-nameservers 8.8.8.8 <------------------------
The call takes about 40 seconds before the callee starts to ring.
If NO DNS is set everything works as it should.
Server Address is: 10.0..
Caller is: 10.0..21
Callee is: 10.0..23
Asterisk Debug on Log WITH DNS:
--- (13 headers 13 lines) ---
Really destroying SIP dialog '29362ddb1510b1787ab1a90c5896e733@10.0.*.*:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.*.21:46983:
OPTIONS sip:2250@10.0.*.21:46983;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK1237646c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as60753440
To: <sip:2250@10.0.*.21:46983;ob>
Contact: <sip:asterisk@10.0.*.*:5060>
Call-ID: 7c275a984c807555320e9aae118d54e2@10.0.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.5.1
Date: Fri, 30 Jan 2015 15:44:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.0.*.21:46983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.*.*:5060;rport=5060;received=10.0.*.*;branch=z9hG4bK1237646c
Call-ID: 7c275a984c807555320e9aae118d54e2@10.0.*.*:5060
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as60753440
To: <sip:2250@10.0.*.21;ob>;tag=z9hG4bK1237646c
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_hltecan-19/r2457
Content-Type: application/sdp
Content-Length: 284
v=0
o=- 3631621547 3631621547 IN IP4 192.168.1.56
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 99 0 8 101
c=IN IP4 192.168.1.56
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (13 headers 13 lines) ---
Really destroying SIP dialog '7c275a984c807555320e9aae118d54e2@10.0.*.*:5060' Method: OPTIONS
<--- SIP read from UDP:10.183.24.74:45793 --->
<------------->
<--- SIP read from UDP:10.183.24.80:33857 --->
<------------->
<--- SIP read from UDP:10.0.*.*5:5060 --->
<------------->
<--- SIP read from UDP:10.183.24.71:52041 --->
<------------->
<--- SIP read from UDP:10.183.24.68:39032 --->
<------------->
<--- SIP read from UDP:10.183.24.79:56344 --->
<------------->
<--- SIP read from UDP:10.0.*.23:5060 --->
<------------->
Reliably Transmitting (no NAT) to 10.0.*.23:5060:
OPTIONS sip:2251@10.0.*.23:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK70e51baa
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as002619f6
To: <sip:2251@10.0.*.23:5060>
Contact: <sip:asterisk@10.0.*.*:5060>
Call-ID: 1bc7c1612b40fc1d775a94776424bfc8@10.0.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.5.1
Date: Fri, 30 Jan 2015 15:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.0.*.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.*.*:5060;received=10.0.*.*;branch=z9hG4bK70e51baa
Call-ID: 1bc7c1612b40fc1d775a94776424bfc8@10.0.*.*:5060
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as002619f6
To: <sip:2251@10.0.*.23>
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: 100rel, norefersub
Allow-Events: presence, refer
User-Agent: Sipek on PJSUA v1.0/win32
Content-Type: application/sdp
Content-Length: 445
v=0
o=- 3631603494 3631603494 IN IP4 10.0.*.23
s=pjmedia
c=IN IP4 10.0.*.23
t=0 0
m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101
a=rtcp:4001 IN IP4 10.0.*.23
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:117 iLBC/8000
a=fmtp:117 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 19 lines) ---
Really destroying SIP dialog '1bc7c1612b40fc1d775a94776424bfc8@10.0.*.*:5060' Method: OPTIONS
Really destroying SIP dialog 'zDL.yzrvaK6RbJf4tBZ2onh.eIi1jLH2' Method: BYE
<--- SIP read from UDP:10.183.24.73:57910 --->
<------------->
Reliably Transmitting (no NAT) to 10.183.24.67:60628:
OPTIONS sip:1702@10.183.24.67:60628;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK376264ab
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as28beccb4
To: <sip:1702@10.183.24.67:60628;ob>
Contact: <sip:asterisk@10.0.*.*:5060>
Call-ID: 38ee61ff676eaa3a55eae3d763f8d57e@10.0.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.5.1
Date: Fri, 30 Jan 2015 15:44:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
asterisk-server-test*CLI> sip set debug off
SIP Debugging Disabled
[Jan 30 10:45:15] NOTICE[32638]: chan_sip.c:27699 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1102
asterisk-server-test*CLI>
Asterisk Debug on Log WITHOUT DNS:
-- (13 headers 13 lines) ---
Really destroying SIP dialog '42d290a25f536cae4a1f07e34166ac78@10.0.*.*:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.*.21:46983:
OPTIONS sip:2250@10.0.*.21:46983;ob SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK0d35e3b6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as74ef6f90
To: <sip:2250@10.0.*.21:46983;ob>
Contact: <sip:asterisk@10.0.*.*:5060>
Call-ID: 11a09f5a147fb1ed3ac38263583e08b4@10.0.*.*:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.5.1
Date: Fri, 30 Jan 2015 16:00:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.0.*.21:46983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.*.*:5060;rport=5060;received=10.0.*.*;branch=z9hG4bK0d35e3b6
Call-ID: 11a09f5a147fb1ed3ac38263583e08b4@10.0.*.*:5060
From: "asterisk" <sip:asterisk@10.0.*.*>;tag=as74ef6f90
To: <sip:2250@10.0.*.21;ob>;tag=z9hG4bK0d35e3b6
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_hltecan-19/r2457
Content-Type: application/sdp
Content-Length: 284
v=0
o=- 3631622477 3631622477 IN IP4 192.168.1.56
s=pjmedia
t=0 0
m=audio 4000 RTP/AVP 99 0 8 101
c=IN IP4 192.168.1.56
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (13 headers 13 lines) ---
Really destroying SIP dialog '11a09f5a147fb1ed3ac38263583e08b4@10.0.*.*:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.*.21:46983 --->
BYE sip:12251@10.0.*.*:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.56:46983;rport;branch=z9hG4bKPjTbmnHnCcHufGuQmpir1muB2gVq2iPXDN
Max-Forwards: 70
From: <sip:2250@10.0.*.*>;tag=Vk7cw3MDLvJIP791VzyGlZz8DD-vQC2z
To: <sip:12251@10.0.*.*>;tag=as503cf33c
Call-ID: ITCMP8tJoFJoM6LmDOuU0UsHTKo83pQL
CSeq: 25722 BYE
User-Agent: CSipSimple_hltecan-19/r2457
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 10.0.*.21:46983 (NAT)
Scheduling destruction of SIP dialog 'ITCMP8tJoFJoM6LmDOuU0UsHTKo83pQL' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 10.0.*.21:46983 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.56:46983;branch=z9hG4bKPjTbmnHnCcHufGuQmpir1muB2gVq2iPXDN;received=10.0.*.21;rport=46983
From: <sip:2250@10.0.*.*>;tag=Vk7cw3MDLvJIP791VzyGlZz8DD-vQC2z
To: <sip:12251@10.0.*.*>;tag=as503cf33c
Call-ID: ITCMP8tJoFJoM6LmDOuU0UsHTKo83pQL
CSeq: 25722 BYE
Server: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
set_destination: Parsing <sip:2251@10.0.*.23:5060> for address/port to send to
set_destination: set destination to 10.0.*.23:5060
Audio is at 15724
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.*.23:5060:
INVITE sip:2251@10.0.*.23:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK5180543c
Max-Forwards: 70
From: <sip:12250@10.0.*.*>;tag=as556be22d
To: <sip:2251@10.0.*.23:5060>;tag=8a00e7db044741dfa676bea72903b0c5
Contact: <sip:12250@10.0.*.*:5060>
Call-ID: 61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060
CSeq: 106 INVITE
User-Agent: Asterisk PBX 11.5.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 1995170150 1995170154 IN IP4 10.0.*.*
s=Asterisk PBX 11.5.1
c=IN IP4 10.0.*.*
t=0 0
m=audio 15724 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Scheduling destruction of SIP dialog '61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060' in 6400 ms (Method: INVITE)
== Spawn extension (default, 12251, 5) exited non-zero on 'SIP/2250-00000002'
<--- SIP read from UDP:10.0.*.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.*.*:5060;received=10.0.*.*;branch=z9hG4bK5180543c
Call-ID: 61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060
From: <sip:12250@10.0.*.*>;tag=as556be22d
To: <sip:2251@10.0.*.23>;tag=8a00e7db044741dfa676bea72903b0c5
CSeq: 106 INVITE
Contact: <sip:2251@10.0.*.23:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: 100rel, norefersub
Content-Type: application/sdp
Content-Length: 248
v=0
o=- 3631604421 3631604420 IN IP4 10.0.*.23
s=pjmedia
c=IN IP4 10.0.*.23
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 0 101
a=rtcp:4003 IN IP4 10.0.*.23
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.*.23:4002
Peer doesn't provide video
set_destination: Parsing <sip:2251@10.0.*.23:5060> for address/port to send to
set_destination: set destination to 10.0.*.23:5060
Transmitting (no NAT) to 10.0.*.23:5060:
ACK sip:2251@10.0.*.23:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK320dae37
Max-Forwards: 70
From: <sip:12250@10.0.*.*>;tag=as556be22d
To: <sip:2251@10.0.*.23:5060>;tag=8a00e7db044741dfa676bea72903b0c5
Contact: <sip:12250@10.0.*.*:5060>
Call-ID: 61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060
CSeq: 106 ACK
User-Agent: Asterisk PBX 11.5.1
Content-Length: 0
---
set_destination: Parsing <sip:2251@10.0.*.23:5060> for address/port to send to
set_destination: set destination to 10.0.*.23:5060
Reliably Transmitting (no NAT) to 10.0.*.23:5060:
BYE sip:2251@10.0.*.23:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.*.*:5060;branch=z9hG4bK79e7dd99
Max-Forwards: 70
From: <sip:12250@10.0.*.*>;tag=as556be22d
To: <sip:2251@10.0.*.23:5060>;tag=8a00e7db044741dfa676bea72903b0c5
Call-ID: 61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060
CSeq: 107 BYE
User-Agent: Asterisk PBX 11.5.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Scheduling destruction of SIP dialog '61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.*.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.*.*:5060;received=10.0.*.*;branch=z9hG4bK79e7dd99
Call-ID: 61ee2eba4b1d323a3ca003e0717c3df6@10.0.*.*:5060
From: <sip:12250@10.0.*.*>;tag=as556be22d
To: <sip:2251@10.0.*.23>;tag=8a00e7db044741dfa676bea72903b0c5
CSeq: 107 BYE
Content-Length: 0
The main difference seems to be the repetitions of SIP READ from other UDP peers when
we look at the differences before the call is correctly placed.
I was wondering what could be the problem.
Thanks