Hello All,
I have been experimenting with Asterisk for a while now. I have a working Asterisk configured on a Virtualbox (Bridged Network) and I have multiple endpoints connected to Asterisk through wireless router (both the host machine and endpoints are on the same wireless network).
This configuration was working fine but recently I have decided to install Asterisk on a dedicated physical server to spice things up a little The new server is connected to the same switch as Wireless Router. and has the IP address: 192.168.1.34 (Please see the diagram below). Endpoints are still on the wireless subnet and their IPs range from 10.0.0.35 - 10.0.0.39. They are connected to Asterisk using the address 192.168.1.34. Although the the SIP signaling is working fine with the new config (Please see the SIP trace below). I can’t hear any voice after I pick up the call. A few things to note, I’m not using a SIP trunk nor am I trying to connect to the internet/public network. All I want is to have a bunch of local softphones that can make phone calls to each other.
I’m using PJSIP and I have tried different combinations of rewrite_contact, force_rport, rtp_symmetric configs but It didn’t solve the issue. I have done some research and I believe I need to make some changes to my rtp.conf as well but not sure what those changes are, currently I’m using out of the box configuration for rtp.conf. I was wondering if someone could help me with this issue.
Thanks,
Kaan
<--- Received SIP request (815 bytes) from UDP:192.168.1.33:64150 --->
INVITE sip:101@192.168.1.34:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:64150;branch=z9hG4bK-524287-1---aa41cbe97bfd7181;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.33:64150;transport=UDP>
To: <sip:101@192.168.1.34:5060>
From: "102"<sip:102@192.168.1.34:5060;transport=UDP>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.16.0
Allow-Events: presence, kpml, talk
Content-Length: 247
v=0
o=Z 60212462 1 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 57784 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
<--- Transmitting SIP response (505 bytes) to UDP:192.168.1.33:64150 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.33:64150;rport=64150;received=192.168.1.33;branch=z9hG4bK-524287-1---aa41cbe97bfd7181
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
To: <sip:101@192.168.1.34>;tag=z9hG4bK-524287-1---aa41cbe97bfd7181
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1637321448/f3a36e342660085e0fd7d9e0673c23ff",opaque="19ce20081653951d",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.8.0
Content-Length: 0
<--- Received SIP request (361 bytes) from UDP:192.168.1.33:64150 --->
ACK sip:101@192.168.1.34:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:64150;branch=z9hG4bK-524287-1---aa41cbe97bfd7181;rport
Max-Forwards: 70
To: <sip:101@192.168.1.34>;tag=z9hG4bK-524287-1---aa41cbe97bfd7181
From: "102"<sip:102@192.168.1.34:5060;transport=UDP>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 1 ACK
Content-Length: 0
<--- Received SIP request (1115 bytes) from UDP:192.168.1.33:64150 --->
INVITE sip:101@192.168.1.34:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:64150;branch=z9hG4bK-524287-1---f37150ff6f9feba8;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.33:64150;transport=UDP>
To: <sip:101@192.168.1.34:5060>
From: "102"<sip:102@192.168.1.34:5060;transport=UDP>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.16.0
Authorization: Digest username="102",realm="asterisk",nonce="1637321448/f3a36e342660085e0fd7d9e0673c23ff",uri="sip:101@192.168.1.34:5060;transport=UDP",response="b4601d921f4edbecb0bf814a612dd5b2",cnonce="3be05850bff930bd82a6c58fed264fe9",nc=00000001,qop=auth,algorithm=md5,opaque="19ce20081653951d"
Allow-Events: presence, kpml, talk
Content-Length: 247
v=0
o=Z 60212462 1 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 57784 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
<--- Transmitting SIP response (313 bytes) to UDP:192.168.1.33:64150 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.33:64150;rport=64150;received=192.168.1.33;branch=z9hG4bK-524287-1---f37150ff6f9feba8
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
To: <sip:101@192.168.1.34>
CSeq: 2 INVITE
Server: Asterisk PBX 18.8.0
Content-Length: 0
-- Executing [101@testing:1] NoOp("PJSIP/102-00000004", "") in new stack
-- Executing [101@testing:2] Dial("PJSIP/102-00000004", "PJSIP/101") in new stack
-- Called PJSIP/101
<--- Transmitting SIP request (949 bytes) to UDP:192.168.1.33:41292 --->
INVITE sip:101@192.168.1.33:41292;rinstance=bc832549eb04f9dd SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj2aaa9ede-e495-431e-ad8c-2562e32e625d
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>
Contact: <sip:asterisk@192.168.1.34:5060>
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13938 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Type: application/sdp
Content-Length: 231
v=0
o=- 9787957 9787957 IN IP4 192.168.1.34
s=Asterisk
c=IN IP4 192.168.1.34
t=0 0
m=audio 17536 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (340 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj2aaa9ede-e495-431e-ad8c-2562e32e625d
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13938 INVITE
Content-Length: 0
<--- Received SIP response (544 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj2aaa9ede-e495-431e-ad8c-2562e32e625d
Contact: <sip:101@192.168.1.33:41292>
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13938 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 0
-- PJSIP/101-00000005 is ringing
<--- Transmitting SIP response (500 bytes) to UDP:192.168.1.33:64150 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.33:64150;rport=64150;received=192.168.1.33;branch=z9hG4bK-524287-1---f37150ff6f9feba8
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
To: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
CSeq: 2 INVITE
Server: Asterisk PBX 18.8.0
Contact: <sip:192.168.1.34:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0
<--- Received SIP response (946 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj2aaa9ede-e495-431e-ad8c-2562e32e625d
Require: timer
Contact: <sip:101@192.168.1.33:41292>
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13938 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 321
v=0
o=Z 0 1 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 47346 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
> 0x7f0f90046d40 -- Strict RTP learning after remote address set to: 192.168.1.33:47346
<--- Transmitting SIP request (417 bytes) to UDP:192.168.1.33:41292 --->
ACK sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj3f4d61a5-e9ce-4e33-9d63-2af23324d125
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13938 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
> 0x7f0f90046d40 -- Strict RTP switching to RTP target address 192.168.1.33:47346 as source
-- PJSIP/101-00000005 answered PJSIP/102-00000004
> 0x7f0f90041350 -- Strict RTP learning after remote address set to: 192.168.1.33:57784
<--- Transmitting SIP response (803 bytes) to UDP:192.168.1.33:64150 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.33:64150;rport=64150;received=192.168.1.33;branch=z9hG4bK-524287-1---f37150ff6f9feba8
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
To: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
CSeq: 2 INVITE
Server: Asterisk PBX 18.8.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.34:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 60212462 3 IN IP4 192.168.1.34
s=Asterisk
c=IN IP4 192.168.1.34
t=0 0
m=audio 18590 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- Channel PJSIP/101-00000005 joined 'simple_bridge' basic-bridge <1a60d8ff-16b0-4bd8-895a-19e149d30584>
-- Channel PJSIP/102-00000004 joined 'simple_bridge' basic-bridge <1a60d8ff-16b0-4bd8-895a-19e149d30584>
> Bridge 1a60d8ff-16b0-4bd8-895a-19e149d30584: switching from simple_bridge technology to native_rtp
> Remotely bridged 'PJSIP/102-00000004' and 'PJSIP/101-00000005' - media will flow directly between them
<--- Transmitting SIP request (949 bytes) to UDP:192.168.1.33:41292 --->
INVITE sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj488b79fe-3054-4193-bcac-f6e64e39137a
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Contact: <sip:asterisk@192.168.1.34:5060>
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13939 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Type: application/sdp
Content-Length: 231
v=0
o=- 9787957 9787958 IN IP4 192.168.1.34
s=Asterisk
c=IN IP4 192.168.1.33
t=0 0
m=audio 57784 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (946 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj488b79fe-3054-4193-bcac-f6e64e39137a
Require: timer
Contact: <sip:101@192.168.1.33:41292>
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13939 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 321
v=0
o=Z 0 2 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 47346 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<--- Transmitting SIP request (417 bytes) to UDP:192.168.1.33:41292 --->
ACK sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPjee0e1b51-d9f9-45de-8658-562459bd24e2
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13939 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
> 0x7f0f90041350 -- Strict RTP switching to RTP target address 192.168.1.33:57784 as source
<--- Received SIP request (410 bytes) from UDP:192.168.1.33:64150 --->
ACK sip:192.168.1.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:64150;branch=z9hG4bK-524287-1---1f5fec7d9c63bd4f;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.33:64150;transport=UDP>
To: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 2 ACK
User-Agent: Zoiper v2.10.16.0
Content-Length: 0
<--- Transmitting SIP request (882 bytes) to UDP:192.168.1.33:64150 --->
INVITE sip:102@192.168.1.33:64150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj52dfdde6-3bc7-413f-90e2-e954b0351b65
From: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
To: "102" <sip:102@192.168.1.34>;tag=2c0a9247
Contact: <sip:192.168.1.34:5060>
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 13835 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Type: application/sdp
Content-Length: 226
v=0
o=- 60212462 4 IN IP4 192.168.1.34
s=Asterisk
c=IN IP4 192.168.1.33
t=0 0
m=audio 47346 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (838 bytes) from UDP:192.168.1.33:64150 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj52dfdde6-3bc7-413f-90e2-e954b0351b65
Require: timer
Contact: <sip:102@192.168.1.33:64150;transport=UDP>
To: "102"<sip:102@192.168.1.34>;tag=2c0a9247
From: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 13835 INVITE
Session-Expires: 1800;refresher=uas
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper v2.10.16.0
Allow-Events: presence, kpml, talk
Content-Length: 240
v=0
o=Z 0 2 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 57784 RTP/AVP 0 3 110 97 8 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<--- Transmitting SIP request (378 bytes) to UDP:192.168.1.33:64150 --->
ACK sip:102@192.168.1.33:64150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPjde3de656-4d40-4e82-9c1a-9a4285139240
From: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
To: "102" <sip:102@192.168.1.34>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 13835 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
<--- Received SIP request (692 bytes) from UDP:192.168.1.33:64150 --->
BYE sip:192.168.1.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:64150;branch=z9hG4bK-524287-1---e29e4c37ac23f79c;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.33:64150;transport=UDP>
To: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
CSeq: 3 BYE
User-Agent: Zoiper v2.10.16.0
Authorization: Digest username="102",realm="asterisk",nonce="1637321448/f3a36e342660085e0fd7d9e0673c23ff",uri="sip:192.168.1.34:5060",response="dfaf9dcd093bcf4e5ad57970e77f4cd3",cnonce="cf637c6b5d7b5659ea8c4fd5c242cd09",nc=00000002,qop=auth,algorithm=md5,opaque="19ce20081653951d"
Content-Length: 0
<--- Transmitting SIP response (347 bytes) to UDP:192.168.1.33:64150 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.33:64150;rport=64150;received=192.168.1.33;branch=z9hG4bK-524287-1---e29e4c37ac23f79c
Call-ID: 62_Y3crl6kulQQpYGYMKlA..
From: "102" <sip:102@192.168.1.34>;tag=2c0a9247
To: <sip:101@192.168.1.34>;tag=b139ac70-4af3-4b1f-859e-852f94bc4446
CSeq: 3 BYE
Server: Asterisk PBX 18.8.0
Content-Length: 0
-- Channel PJSIP/102-00000004 left 'native_rtp' basic-bridge <1a60d8ff-16b0-4bd8-895a-19e149d30584>
-- Channel PJSIP/101-00000005 left 'native_rtp' basic-bridge <1a60d8ff-16b0-4bd8-895a-19e149d30584>
== Spawn extension (testing, 101, 2) exited non-zero on 'PJSIP/102-00000004'
<--- Transmitting SIP request (949 bytes) to UDP:192.168.1.33:41292 --->
INVITE sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj5742b99c-4b6c-4e42-9ab4-feafdf3df2ab
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Contact: <sip:asterisk@192.168.1.34:5060>
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13940 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800;refresher=uac
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Type: application/sdp
Content-Length: 231
v=0
o=- 9787957 9787959 IN IP4 192.168.1.34
s=Asterisk
c=IN IP4 192.168.1.34
t=0 0
m=audio 17536 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (946 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj5742b99c-4b6c-4e42-9ab4-feafdf3df2ab
Require: timer
Contact: <sip:101@192.168.1.33:41292>
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13940 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.8 v2.10.17.2
Allow-Events: presence, kpml, talk
Content-Length: 321
v=0
o=Z 0 3 IN IP4 192.168.1.33
s=Z
c=IN IP4 192.168.1.33
t=0 0
m=audio 47346 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
<--- Transmitting SIP request (417 bytes) to UDP:192.168.1.33:41292 --->
ACK sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPjd9ec7ddd-fdff-4e0d-b845-283856c31c6b
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13940 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
<--- Transmitting SIP request (417 bytes) to UDP:192.168.1.33:41292 --->
BYE sip:101@192.168.1.33:41292 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;rport;branch=z9hG4bKPj6c075731-7e97-4c91-a196-690bd2dc3114
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13941 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
<--- Received SIP response (417 bytes) from UDP:192.168.1.33:41292 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;rport=5060;branch=z9hG4bKPj6c075731-7e97-4c91-a196-690bd2dc3114
Contact: <sip:101@192.168.1.33:41292>
To: <sip:101@192.168.1.33;rinstance=bc832549eb04f9dd>;tag=88d9e745
From: "102" <sip:102@192.168.1.34>;tag=c2e804ba-d0be-4b0e-996f-0bafa94f760e
Call-ID: 8cb28471-70ff-4a8f-921a-b03f0498f41f
CSeq: 13941 BYE
User-Agent: Z 5.5.8 v2.10.17.2
Content-Length: 0