SIP-session established... RTP-flow not

I can make call, but the other end does not hear me. So problem with RTP-flow…

Can someone guide me to the solution ?

On the Asterisk-server I have this (iptables):

-A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited

In rtp.conf I have this :

rtpstart=11000
rtpend=11500

Asterisk is behind firewall. Endian firewall has following configuration :

enable SIP proxy transparant
RTP port low : 11000
RTP port high : 11500

Firewall port forwarding : uplink:5060 >>> asterisk_ip:5060

Asterisk himself says :

-- Executing [050510484@intern:1] NoOp("SIP/grandstream-09813b58", "via 3StarsNet") in new stack
-- Executing [050510484@intern:2] Dial("SIP/grandstream-09813b58", "SIP/3starsnet/050510484") in new stack
-- Called 3starsnet/050510484
-- SIP/3starsnet-0981bf08 is making progress passing it to SIP/grandstream-09813b58
-- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58

== Spawn extension (intern, 050510484, 2) exited non-zero on ‘SIP/grandstream-09813b58’

What do I need in sip.conf to overcome these rtp-problems ??
I have :
externip=78.21.62.99
canreinvite=no
jbenable = yes

[3starsnet]
type=peer

nat=yes

Thanks for the help !

My setup is maybe to specific to advise…

But can someone tell me some rules I need to take into account when employing an Asterisk-server behind a firewall…

Meaning : how to resolv RTP-issues ?
SIP is no problem, but there is no audio…

Thanks !

UPDATE :

I have forwarded a range of RTP-ports (UDP 11000:11500) from my firewall to Asterisk and when a call comes in I can hear the other side…
But the other side cannot hear me.
I have opened the same range of RTP-ports (UDP) on the firewall for outgoing traffic, but with no luck.

-- Executing [0494456661@intern:2] Dial("SIP/grandstream-099a21a0", "SIP/3starsnet/0494456661") in new stack

Audio is at 78.21.62.99 port 11436
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 85.119.188.3:5060:
INVITE sip:0494456661@85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK7b26740a;rport
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3
Contact: sip:092779077@78.21.62.99
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 19:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 5535 5535 IN IP4 78.21.62.99
s=session
c=IN IP4 78.21.62.99
t=0 0
m=audio 11436 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Called 3starsnet/0494456661

asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK7b26740a;rport=5060
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3:5060;tag=470f6df9d438ef0e611aed43a3c90fcf.b1d5
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“sip.3starsnet.com”, nonce="4a4523fe0001646dd2e3ca232cf01d3421fa0968953176da"
Server: Enswitch SIP proxy
Content-Length: 0
Warning: 392 85.119.188.3:5060 “Noisy feedback tells: pid=6614 req_src_ip=78.21.62.99 req_src_port=5060 in_uri=sip:0494456661@85.119.188.3:5060 out_uri=sip:0494456661@85.119.188.3:5060 via_cnt==1”

<------------->
— (10 headers 0 lines) —
Transmitting (no NAT) to 85.119.188.3:5060:
ACK sip:0494456661@85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK7b26740a;rport
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3;tag=470f6df9d438ef0e611aed43a3c90fcf.b1d5
Contact: sip:092779077@78.21.62.99
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Audio is at 78.21.62.99 port 11436
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 85.119.188.3:5060:
INVITE sip:0494456661@85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK7fd05d33;rport
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3
Contact: sip:092779077@78.21.62.99
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=“092779077”, realm=“sip.3starsnet.com”, algorithm=MD5, uri="sip:0494456661@85.119.188.3", nonce=“4a4523fe0001646dd2e3ca232cf01d3421fa0968953176da”, response="4373d2bf78fc57cdf0780ba1212c3838"
Date: Fri, 26 Jun 2009 19:38:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 5535 5536 IN IP4 78.21.62.99
s=session
c=IN IP4 78.21.62.99
t=0 0
m=audio 11436 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK7fd05d33;rport=5060
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3:5060
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 103 INVITE
Server: Enswitch SIP proxy
Content-Length: 0
Warning: 392 85.119.188.3:5060 “Noisy feedback tells: pid=6615 req_src_ip=78.21.62.99 req_src_port=5060 in_uri=sip:0494456661@85.119.188.3:5060 out_uri=sip:0494456661@85.119.188.31:5060;transport=udp via_cnt==1”

<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.2:5060;received=78.21.62.99;branch=z9hG4bK7fd05d33;rport=5060
Record-Route: sip:85.119.188.3;lr=on;ftag=as26c76e39
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3:5060;tag=as400a1920
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 103 INVITE
User-Agent: Integrics Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:0494456661@85.119.188.31:5060
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 4665 4665 IN IP4 85.119.188.3
s=session
c=IN IP4 85.119.188.3
t=0 0
m=audio 14042 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=direction:active

<------------->
— (13 headers 14 lines) —
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 85.119.188.3:14042
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xa (gsm|alaw), peer - audio=0xa (gsm|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 85.119.188.3:14042
– SIP/3starsnet-099a8a88 is making progress passing it to SIP/grandstream-099a21a0
Audio is at 192.168.1.248 port 11214
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKfa39d9dd56b45ba0;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=3b0a2e0a23d8371b
To: sip:0494456661@192.168.1.248;tag=as1a7d83eb
Call-ID: 9dd5441696037a8e@192.168.1.13
CSeq: 7307 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:0494456661@192.168.1.248
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 5535 5535 IN IP4 192.168.1.248
s=session
c=IN IP4 192.168.1.248
t=0 0
m=audio 11214 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Reliably Transmitting (no NAT) to 85.119.188.3:5060:
OPTIONS sip:85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK0d740116;rport
From: “asterisk” sip:asterisk@78.21.62.99;tag=as23d95dde
To: sip:85.119.188.3
Contact: sip:asterisk@78.21.62.99
Call-ID: 1e3888032c50bc894baab7fc0efebc8a@78.21.62.99
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 26 Jun 2009 19:38:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK0d740116;rport=5060
From: “asterisk” sip:asterisk@192.168.2.2:5060;tag=as23d95dde
To: sip:85.119.188.3;tag=470f6df9d438ef0e611aed43a3c90fcf.9aac
Call-ID: 1e3888032c50bc894baab7fc0efebc8a@78.21.62.99
CSeq: 102 OPTIONS
Server: Enswitch SIP proxy
Content-Length: 0
Warning: 392 85.119.188.3:5060 “Noisy feedback tells: pid=6618 req_src_ip=78.21.62.99 req_src_port=5060 in_uri=sip:85.119.188.3 out_uri=sip:85.119.188.3 via_cnt==1”

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘1e3888032c50bc894baab7fc0efebc8a@78.21.62.99’ Method: OPTIONS
[Jun 26 21:38:50] NOTICE[5570]: chan_sip.c:7653 sip_reregister: – Re-registration for 092779077@85.119.188.3
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 85.119.188.3:5060:
REGISTER sip:85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK39655b53;rport
From: sip:092779077@85.119.188.3;tag=as520f110b
To: sip:092779077@85.119.188.3
Call-ID: 64751f9a1c77d59d5da2c4731cc8aefe@127.0.0.1
CSeq: 158 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“092779077”, realm=“85.119.188.3”, algorithm=MD5, uri=“sip:85.119.188.3”, nonce=“4a4523a70001577060fbd7639beff6313a8d82f1f258ed74”, response="b438e03712720202a22546b5db4837ba"
Expires: 120
Contact: sip:s@78.21.62.99
Event: registration
Content-Length: 0


asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK39655b53;rport=5060
From: sip:092779077@85.119.188.3:5060;tag=as520f110b
To: sip:092779077@85.119.188.3:5060;tag=470f6df9d438ef0e611aed43a3c90fcf.b812
Call-ID: 64751f9a1c77d59d5da2c4731cc8aefe@127.0.0.1
CSeq: 158 REGISTER
WWW-Authenticate: Digest realm=“85.119.188.3”, nonce=“4a45241000016706247719dca6add8457ca06d73c923bc1d”, stale=true
Server: Enswitch SIP proxy
Content-Length: 0
Warning: 392 85.119.188.3:5060 “Noisy feedback tells: pid=6620 req_src_ip=78.21.62.99 req_src_port=5060 in_uri=sip:85.119.188.3:5060 out_uri=sip:85.119.188.3:5060 via_cnt==1”

<------------->
— (10 headers 0 lines) —
Responding to challenge, registration to domain/host name 85.119.188.3
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 85.119.188.3:5060:
REGISTER sip:85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK6f935189;rport
From: sip:092779077@85.119.188.3;tag=as099dcba0
To: sip:092779077@85.119.188.3
Call-ID: 64751f9a1c77d59d5da2c4731cc8aefe@127.0.0.1
CSeq: 159 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“092779077”, realm=“85.119.188.3”, algorithm=MD5, uri=“sip:85.119.188.3”, nonce=“4a45241000016706247719dca6add8457ca06d73c923bc1d”, response="554efc4cbad760978a902a12abfb9710"
Expires: 120
Contact: sip:s@78.21.62.99
Event: registration
Content-Length: 0


asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6f935189;rport=5060
From: sip:092779077@85.119.188.3:5060;tag=as099dcba0
To: sip:092779077@85.119.188.3:5060;tag=470f6df9d438ef0e611aed43a3c90fcf.568a
Call-ID: 64751f9a1c77d59d5da2c4731cc8aefe@127.0.0.1
CSeq: 159 REGISTER
Contact: sip:s@192.168.2.2:5060;expires=120
Server: Enswitch SIP proxy
Content-Length: 0
Warning: 392 85.119.188.3:5060 “Noisy feedback tells: pid=6620 req_src_ip=78.21.62.99 req_src_port=5060 in_uri=sip:85.119.188.3:5060 out_uri=sip:85.119.188.3:5060 via_cnt==1”

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘64751f9a1c77d59d5da2c4731cc8aefe@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Jun 26 21:38:50] NOTICE[5570]: chan_sip.c:12942 handle_response_register: Outbound Registration: Expiry for 85.119.188.3 is 120 sec (Scheduling reregistration in 105 s)
asterisk*CLI>
<— SIP read from 85.119.188.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2:5060;received=78.21.62.99;branch=z9hG4bK7fd05d33;rport=5060
Record-Route: sip:85.119.188.3;lr=on;ftag=as26c76e39
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3:5060;tag=as400a1920
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 103 INVITE
User-Agent: Integrics Enswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
upported: replaces
Contact: sip:0494456661@85.119.188.31:5060
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 4665 4666 IN IP4 85.119.188.3
s=session
c=IN IP4 85.119.188.3
t=0 0
m=audio 14042 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=direction:active

<------------->
— (13 headers 14 lines) —
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 85.119.188.3:14042
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xa (gsm|alaw), peer - audio=0xa (gsm|alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 85.119.188.3:14042
list_route: hop: sip:85.119.188.3;lr=on;ftag=as26c76e39
set_destination: Parsing sip:85.119.188.3;lr=on;ftag=as26c76e39 for address/port to send to
set_destination: set destination to 85.119.188.3, port 5060
Transmitting (no NAT) to 85.119.188.3:5060:
ACK sip:0494456661@85.119.188.31:5060 SIP/2.0
Via: SIP/2.0/UDP 78.21.62.99:5060;branch=z9hG4bK3e918599;rport
Route: sip:85.119.188.3;lr=on;ftag=as26c76e39
From: “grandstream” sip:092779077@sip.3starsnet.com;tag=as26c76e39
To: sip:0494456661@85.119.188.3;tag=as400a1920
Contact: sip:092779077@78.21.62.99
Call-ID: 0a1f9f0767f9a73446f7dfad4bcba369@sip.3starsnet.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


-- SIP/3starsnet-099a8a88 answered SIP/grandstream-099a21a0

Audio is at 192.168.1.248 port 11214
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKfa39d9dd56b45ba0;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=3b0a2e0a23d8371b
To: sip:0494456661@192.168.1.248;tag=as1a7d83eb
Call-ID: 9dd5441696037a8e@192.168.1.13
CSeq: 7307 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:0494456661@192.168.1.248
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 5535 5536 IN IP4 192.168.1.248
s=session
c=IN IP4 192.168.1.248
t=0 0
m=audio 11214 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Packet2Packet bridging SIP/grandstream-099a21a0 and SIP/3starsnet-099a8a88
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:0494456661@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKda943fe64bc64037
From: sip:grandstream@192.168.1.248;tag=3b0a2e0a23d8371b
To: sip:0494456661@192.168.1.248;tag=as1a7d83eb
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:0494456661@192.168.1.248", nonce=“165e1203”, response="e484c0f59c56c6f9848ed24e76202f28"
Call-ID: 9dd5441696037a8e@192.168.1.13
CSeq: 7307 ACK
User-Agent: Grandstream GXP2010 1.2.1.4
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
asterisk*CLI> exit
<— SIP read from 192.168.1.13:5060 —>
BYE sip:0494456661@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb0aee30a69f907be
From: sip:grandstream@192.168.1.248;tag=3b0a2e0a23d8371b
To: sip:0494456661@192.168.1.248;tag=as1a7d83eb
Supported: path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:0494456661@192.168.1.248", nonce=“165e1203”, response="fb6d3a4eca6dc1ca57e330e744bab9cc"
Call-ID: 9dd5441696037a8e@192.168.1.13
CSeq: 7308 BYEexit
User-Agent: Grandstream GXP2010 1.2.1.4
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb0aee30a69f907be;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=3b0a2e0a23d8371b
To: sip:0494456661@192.168.1.248;tag=as1a7d83eb
Call-ID: 9dd5441696037a8e@192.168.1.13
CSeq: 7308 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0