Asterisk 16 brand new install cant not hear sound between 2 calls

sean · October 19, 2022, 4:01pm

I install asterisk 16 and registered 2 sip accounts however when i can from one ext to other, there is no sound at all.
Here is the sip debug, any help would be much appreciated.

== Using SIP RTP CoS mark 5
> 0x7f708671af00 – Strict RTP learning after remote address set to: 142.183.119.36:37948
– Executing [25@ShabOut:1] Set(“SIP/26-00000000”, “CALLERID(all)=“The Visa” <416-477-2545>”) in new stack
– Executing [25@ShabOut:2] Answer(“SIP/26-00000000”, “10”) in new stack
– Executing [25@ShabOut:3] Dial(“SIP/26-00000000”, “SIP/25”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/25
– SIP/25-00000001 is ringing
> 0x7f70c4008600 – Strict RTP learning after remote address set to: 192.168.10.136:5022
– SIP/25-00000001 answered SIP/26-00000000
– Channel SIP/25-00000001 joined ‘simple_bridge’ basic-bridge <819a67e6-43bc-4842-a4c2-f00452d8bb5a>
– Channel SIP/26-00000000 joined ‘simple_bridge’ basic-bridge <819a67e6-43bc-4842-a4c2-f00452d8bb5a>
> Bridge 819a67e6-43bc-4842-a4c2-f00452d8bb5a: switching from simple_bridge technology to native_rtp
> Locally RTP bridged ‘SIP/26-00000000’ and ‘SIP/25-00000001’ in stack
ne-r115-291cl*CLI> sip set debug on
SIP Debugging enabled

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0a5e6172
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 181

v=0
o=25 8005 8000 IN IP4 192.168.10.136
s=SIP Call
c=IN IP4 192.168.10.136
t=0 0
m=audio 5022 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
<------------->
— (12 headers 10 lines) —
set_destination: Parsing sip:25@192.168.10.136:5070 for address/port to send to
set_destination: set destination to 192.168.10.136:5070
Transmitting (no NAT) to 192.168.10.136:5070:
ACK sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK37f28cee
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.28.0
Content-Length: 0

Reliably Transmitting (no NAT) to 142.183.119.36:23557:
OPTIONS sip:26@142.183.119.36:59722 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK7e88440c
Max-Forwards: 70
From: “asterisk” sip:asterisk@184.107.85.181;tag=as5b791bbe
To: sip:26@142.183.119.36:59722
Contact: sip:asterisk@184.107.85.181:5060
Call-ID: 7c61f38f7bca4f131c5a72d574ad4018@184.107.85.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.28.0
Date: Wed, 19 Oct 2022 15:10:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK7e88440c
From: “asterisk” sip:asterisk@184.107.85.181;tag=as5b791bbe
To: sip:26@142.183.119.36:59722;tag=148534160
Call-ID: 7c61f38f7bca4f131c5a72d574ad4018@184.107.85.181:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘7c61f38f7bca4f131c5a72d574ad4018@184.107.85.181:5060’ Method: OPTIONS
Really destroying SIP dialog ‘181395524-5070-1@BJC.BGI.BA.BDG’ Method: REGISTER

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0a5e6172
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 181

v=0
o=25 8005 8000 IN IP4 192.168.10.136
s=SIP Call
c=IN IP4 192.168.10.136
t=0 0
m=audio 5022 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
<------------->
— (12 headers 10 lines) —
set_destination: Parsing sip:25@192.168.10.136:5070 for address/port to send to
set_destination: set destination to 192.168.10.136:5070
Transmitting (no NAT) to 192.168.10.136:5070:
ACK sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK3066f618
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.28.0
Content-Length: 0

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0a5e6172
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 181

v=0
o=25 8005 8000 IN IP4 192.168.10.136
s=SIP Call
c=IN IP4 192.168.10.136
t=0 0
m=audio 5022 RTP/AVP 0 8
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
<------------->
— (12 headers 10 lines) —
set_destination: Parsing sip:25@192.168.10.136:5070 for address/port to send to
set_destination: set destination to 192.168.10.136:5070
Transmitting (no NAT) to 192.168.10.136:5070:
ACK sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK296cb524
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
To: sip:25@192.168.10.136:5070;tag=582088681
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.28.0
Content-Length: 0

<— SIP read from UDP:129.222.184.158:43612 —>
BYE sip:4164772545@184.107.85.181:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.136:5070;branch=z9hG4bK2147333033;rport
From: sip:25@192.168.10.136:5070;tag=582088681
To: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 103 BYE
Contact: sip:25@192.168.10.136:5070
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 129.222.184.158:43612 (no NAT)
Scheduling destruction of SIP dialog ‘227733f748882bcf59d46ab30ef94117@184.107.85.181:5060’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.136:5070;branch=z9hG4bK2147333033;received=129.222.184.158;rport=43612
From: sip:25@192.168.10.136:5070;tag=582088681
To: “The Visa” sip:4164772545@184.107.85.181;tag=as4295c201
Call-ID: 227733f748882bcf59d46ab30ef94117@184.107.85.181:5060
CSeq: 103 BYE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/25-00000001 left ‘native_rtp’ basic-bridge <819a67e6-43bc-4842-a4c2-f00452d8bb5a>
– Channel SIP/26-00000000 left ‘native_rtp’ basic-bridge <819a67e6-43bc-4842-a4c2-f00452d8bb5a>
== Spawn extension (ShabOut, 25, 3) exited non-zero on ‘SIP/26-00000000’
Scheduling destruction of SIP dialog ‘1009733073-5064-25@BJC.BGI.BA.BBE’ in 9856 ms (Method: ACK)
set_destination: Parsing sip:26@142.183.119.36:59722 for address/port to send to
set_destination: set destination to 142.183.119.36:59722
Reliably Transmitting (no NAT) to 142.183.119.36:59722:
BYE sip:26@142.183.119.36:59722 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK2c780123;rport
Max-Forwards: 70
From: sip:25@184.107.85.181;tag=as5a3f2195
To: “26” sip:26@184.107.85.181;tag=1564295546
Call-ID: 1009733073-5064-25@BJC.BGI.BA.BBE
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.28.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK2c780123;rport=5060
From: sip:25@184.107.85.181;tag=as5a3f2195
To: “26” sip:26@184.107.85.181;tag=1564295546
Call-ID: 1009733073-5064-25@BJC.BGI.BA.BBE
CSeq: 102 BYE
Contact: sip:26@142.183.119.36:59722
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘1009733073-5064-25@BJC.BGI.BA.BBE’ Method: ACK
Really destroying SIP dialog ‘227733f748882bcf59d46ab30ef94117@184.107.85.181:5060’ Method: BYE

<— SIP read from UDP:142.183.119.36:23557 —>

<------------->
ne-r115-291cl*CLI> sip set debug off
SIP Debugging Disabled

david551 · October 19, 2022, 5:56pm

The log is incomplete. It is missing the whole of the incoming INVITE transaction and the request from the outgoing one. Also, please you the full log, which has time stamps, not a screen scrape of the console.

sean · October 19, 2022, 6:27pm

this is the entire log, however it does not show timestamp i used sip set debug on.
Now i made some changes to sip.con and ext 25 can hear it but not 26.

ne-r115-291clCLI>
ne-r115-291clCLI>
ne-r115-291clCLI>
ne-r115-291clCLI>
Really destroying SIP dialog ‘1298330281-5070-89@BJC.BGI.BA.BDG’ Method: OPTIONS

<— SIP read from UDP:129.222.184.158:43612 —>
OPTIONS sip:184.107.85.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.136:5070;branch=z9hG4bK921523081;rport
From: sip:25@184.107.85.181;tag=818169877
To: sip:184.107.85.181
Call-ID: 1995483615-5070-90@BJC.BGI.BA.BDG
CSeq: 890 OPTIONS
Contact: sip:25@192.168.10.136:5070
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2160 1.0.11.10
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 129.222.184.158:43612 (NAT)
Looking for s in public (domain 184.107.85.181)

<— Transmitting (NAT) to 129.222.184.158:43612 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.136:5070;branch=z9hG4bK921523081;received=129.222.184.158;rport=43612
From: sip:25@184.107.85.181;tag=818169877
To: sip:184.107.85.181;tag=as3a73187d
Call-ID: 1995483615-5070-90@BJC.BGI.BA.BDG
CSeq: 890 OPTIONS
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1995483615-5070-90@BJC.BGI.BA.BDG’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:142.183.119.36:23557 —>
INVITE sip:25@184.107.85.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK487871944;rport
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1000 INVITE
Contact: “26” sip:26@142.183.119.36:59722
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.11.35
Privacy: none
P-Preferred-Identity: “26” sip:26@184.107.85.181
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=3C-8C-F8-F9-98-15
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-3E-EF-06
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:465

v=0
o=26 8002 8000 IN IP4 142.183.119.36
s=SIP Call
i=(o=IN IP4 192.168.10.114)
c=IN IP4 142.183.119.36
t=0 0
m=audio 27928 RTP/AVP 0 8 18 2 4 9 97 123 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:123 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (19 headers 21 lines) —
Sending to 142.183.119.36:23557 (NAT)
Sending to 142.183.119.36:23557 (NAT)
Using INVITE request as basis request - 224910958-5064-101@BJC.BGI.BA.BBE
Found peer ‘26’ for ‘26’ from 142.183.119.36:23557
== Using SIP RTP CoS mark 5
Got SDP version 8000 and unique parts [26 8002 IN IP4 142.183.119.36]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 123
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 G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format opus for ID 123
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729|codec2|g723|gsm|g726|g726aal2|adpcm|slin|slin12|slin16|slin24|slin32|slin44|slin48|slin96|slin192|lpc10|speex|speex16|speex32|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|h265|mpeg4|vp8|vp9|red|t140|t38|silk8|silk12|silk16|silk24), peer - audio=(ulaw|g726|g723|alaw|g722|g729|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729|g723|g726|ilbc|g722|opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f70849367b0 – Strict RTP learning after remote address set to: 142.183.119.36:27928
Peer audio RTP is at port 142.183.119.36:27928
Looking for 25 in ShabOut (domain 184.107.85.181)
sip_route_dump: route/path hop: sip:26@142.183.119.36:59722

<— Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK487871944;received=142.183.119.36;rport=23557
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1000 INVITE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:25@184.107.85.181:5060
Content-Length: 0

<------------>
– Executing [25@ShabOut:1] Set(“SIP/26-00000042”, “CALLERID(all)=“The Visa” <416-477-2545>”) in new stack
– Executing [25@ShabOut:2] Answer(“SIP/26-00000042”, “10”) in new stack
Audio is at 11314
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK487871944;received=142.183.119.36;rport=23557
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181;tag=as53c56adb
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1000 INVITE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:25@184.107.85.181:5060
Content-Type: application/sdp
Require: timer
Content-Length: 485

v=0
o=root 1893352526 1893352526 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 184.107.85.181
t=0 0
m=audio 11314 RTP/AVP 0 8 18 4 2 97 9 123 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:9 G722/8000
a=rtpmap:123 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv

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

<— SIP read from UDP:142.183.119.36:23557 —>
ACK sip:25@184.107.85.181:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK1070638546;rport
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181;tag=as53c56adb
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1000 ACK
Contact: sip:26@142.183.119.36:59722
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
> 0x7f70849367b0 – Strict RTP qualifying stream type: audio
> 0x7f70849367b0 – Strict RTP switching source address to 142.183.119.36:23984
– Executing [25@ShabOut:3] Dial(“SIP/26-00000042”, “SIP/25”) in new stack
== Using SIP RTP CoS mark 5
We think we can do text
Audio is at 14616
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin12 to SDP
Adding codec slin16 to SDP
Adding codec slin24 to SDP
Adding codec slin32 to SDP
Adding codec slin44 to SDP
Adding codec slin48 to SDP
Adding codec slin96 to SDP
Adding codec slin192 to SDP
Adding codec lpc10 to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec testlaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK1160d8c6;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.28.0
Date: Wed, 19 Oct 2022 18:20:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 715

v=0
o=root 1872355781 1872355781 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 184.107.85.181
t=0 0
m=audio 14616 RTP/AVP 0 8 3 2 112 5 10 122 118 98 124 125 126 127 96 7 97 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:122 L16/12000
a=rtpmap:118 L16/16000
a=rtpmap:98 L16/24000
a=rtpmap:124 L16/32000
a=rtpmap:125 L16/44000
a=rtpmap:126 L16/48000
a=rtpmap:127 L16/96000
a=rtpmap:96 L16/192000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

-- Called SIP/25

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK1160d8c6;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK1160d8c6;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 102 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
sip_route_dump: route/path hop: sip:25@192.168.10.136:5070
– SIP/25-00000043 is ringing
Really destroying SIP dialog ‘2092304622-5070-1@BJC.BGI.BA.BDG’ Method: REGISTER

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK1160d8c6;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 102 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 333

-- SIP/25-00000043 answered SIP/26-00000042
-- Channel SIP/25-00000043 joined 'simple_bridge' basic-bridge <b5c433cf-b738-4c33-bedc-c53ce5d0c020>
-- Channel SIP/26-00000042 joined 'simple_bridge' basic-bridge <b5c433cf-b738-4c33-bedc-c53ce5d0c020>
   > Bridge b5c433cf-b738-4c33-bedc-c53ce5d0c020: switching from simple_bridge technology to native_rtp

Audio is at 11314
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 142.183.119.36:23557:
INVITE sip:26@142.183.119.36:59722 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK26e89a1c;rport
Max-Forwards: 70
From: sip:25@184.107.85.181;tag=as53c56adb
To: “26” sip:26@184.107.85.181;tag=2038763965
Contact: sip:25@184.107.85.181:5060
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.28.0
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 1893352526 1893352527 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 192.168.10.136
t=0 0
m=audio 5038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

Audio is at 14616
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK060564ef;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1872355781 1872355782 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 142.183.119.36
t=0 0
m=audio 27928 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

   > Remotely bridged 'SIP/26-00000042' and 'SIP/25-00000043' - media will flow directly between them

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK26e89a1c;rport=5060
From: sip:25@184.107.85.181;tag=as53c56adb
To: “26” sip:26@184.107.85.181;tag=2038763965
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 102 INVITE
Contact: sip:26@142.183.119.36:59722
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK26e89a1c;rport=5060
From: sip:25@184.107.85.181;tag=as53c56adb
To: “26” sip:26@184.107.85.181;tag=2038763965
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 102 INVITE
Contact: sip:26@142.183.119.36:59722
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp

sean · October 19, 2022, 6:27pm

Content-Length:243

Retransmitting #1 (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK060564ef;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1872355781 1872355782 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 142.183.119.36
t=0 0
m=audio 27928 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK060564ef;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 103 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK060564ef;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 103 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 213

Audio is at 14616
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0204595a;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1872355781 1872355783 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 142.183.119.36
t=0 0
m=audio 52026 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

Retransmitting #1 (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0204595a;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1872355781 1872355783 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 142.183.119.36
t=0 0
m=audio 52026 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

   > 0x7f70b0013cd0 -- Strict RTP qualifying stream type: audio

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0204595a;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 104 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK0204595a;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 104 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 213

<— SIP read from UDP:142.183.119.36:23557 —>
BYE sip:25@184.107.85.181:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK1992029372;rport
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181;tag=as53c56adb
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1001 BYE
Contact: sip:26@142.183.119.36:59722
X-Grandstream-PBX: true
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 142.183.119.36:23557 (NAT)
Scheduling destruction of SIP dialog ‘224910958-5064-101@BJC.BGI.BA.BBE’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK1992029372;received=142.183.119.36;rport=23557
From: “26” sip:26@184.107.85.181;tag=2038763965
To: sip:25@184.107.85.181;tag=as53c56adb
Call-ID: 224910958-5064-101@BJC.BGI.BA.BBE
CSeq: 1001 BYE
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/26-00000042 left ‘native_rtp’ basic-bridge
Audio is at 14616
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 129.222.184.158:43612:
INVITE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK615d44ba;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Contact: sip:4164772545@184.107.85.181:5060
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1872355781 1872355784 IN IP4 184.107.85.181
s=Asterisk PBX 16.28.0
c=IN IP4 184.107.85.181
t=0 0
m=audio 14616 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

-- Channel SIP/25-00000043 left 'native_rtp' basic-bridge <b5c433cf-b738-4c33-bedc-c53ce5d0c020>

== Spawn extension (ShabOut, 25, 3) exited non-zero on ‘SIP/26-00000042’
Scheduling destruction of SIP dialog ‘1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK615d44ba;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 105 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

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

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK615d44ba;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 105 INVITE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 213

Reliably Transmitting (NAT) to 129.222.184.158:43612:
BYE sip:25@192.168.10.136:5070 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK43233b97;rport
Max-Forwards: 70
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 106 BYE
User-Agent: Asterisk PBX 16.28.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Scheduling destruction of SIP dialog ‘1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:129.222.184.158:43612 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK43233b97;rport=5060
From: “The Visa” sip:4164772545@184.107.85.181;tag=as4262650c
To: sip:25@192.168.10.136:5070;tag=1291737934
Call-ID: 1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060
CSeq: 106 BYE
Contact: sip:25@192.168.10.136:5070
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘1787a04a5bfe88fd403cd84c36c1d3f0@184.107.85.181:5060’ Method: INVITE
Really destroying SIP dialog ‘224910958-5064-101@BJC.BGI.BA.BBE’ Method: BYE
Reliably Transmitting (NAT) to 142.183.119.36:23557:
OPTIONS sip:26@142.183.119.36:59722 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK4550477b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@184.107.85.181;tag=as37b838c8
To: sip:26@142.183.119.36:59722
Contact: sip:asterisk@184.107.85.181:5060
Call-ID: 65d657cc7b595626274adfc80954ea90@184.107.85.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.28.0
Date: Wed, 19 Oct 2022 18:20:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK4550477b;rport=5060
From: “asterisk” sip:asterisk@184.107.85.181;tag=as37b838c8
To: sip:26@142.183.119.36:59722;tag=29484826
Call-ID: 65d657cc7b595626274adfc80954ea90@184.107.85.181:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘65d657cc7b595626274adfc80954ea90@184.107.85.181:5060’ Method: OPTIONS

<— SIP read from UDP:142.183.119.36:23557 —>
OPTIONS sip:184.107.85.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK303326670;rport
From: sip:26@184.107.85.181;tag=905991329
To: sip:184.107.85.181
Call-ID: 842498504-5064-102@BJC.BGI.BA.BBE
CSeq: 1010 OPTIONS
Contact: sip:26@142.183.119.36:59722
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.11.35
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 142.183.119.36:23557 (NAT)
Looking for s in public (domain 184.107.85.181)

<— Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK303326670;received=142.183.119.36;rport=23557
From: sip:26@184.107.85.181;tag=905991329
To: sip:184.107.85.181;tag=as0adb3240
Call-ID: 842498504-5064-102@BJC.BGI.BA.BBE
CSeq: 1010 OPTIONS
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘842498504-5064-102@BJC.BGI.BA.BBE’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog ‘1942714876-5064-100@BJC.BGI.BA.BBE’ Method: OPTIONS

<— SIP read from UDP:142.183.119.36:23557 —>
REGISTER sip:184.107.85.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK692238431;rport
From: sip:26@184.107.85.181;tag=181936273
To: sip:26@184.107.85.181
Call-ID: 11532509-5064-1@BJC.BGI.BA.BBE
CSeq: 2044 REGISTER
Contact: sip:26@142.183.119.36:59722;reg-id=3;+sip.instance=“urn:uuid:00000000-0000-1000-8000-C074AD3EEF06”
Authorization: Digest username=“26”, realm=“asterisk”, nonce=“10924af9”, uri=“sip:184.107.85.181”, response=“f73fb8768353138bd4ff8e1f711c403b”, algorithm=MD5
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.11.35
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 142.183.119.36:23557 (NAT)
Sending to 142.183.119.36:23557 (NAT)

<— Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK692238431;received=142.183.119.36;rport=23557
From: sip:26@184.107.85.181;tag=181936273
To: sip:26@184.107.85.181;tag=as65d43bad
Call-ID: 11532509-5064-1@BJC.BGI.BA.BBE
CSeq: 2044 REGISTER
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“14da8f7a”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘11532509-5064-1@BJC.BGI.BA.BBE’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:142.183.119.36:23557 —>
REGISTER sip:184.107.85.181 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK1474822169;rport
From: sip:26@184.107.85.181;tag=181936273
To: sip:26@184.107.85.181
Call-ID: 11532509-5064-1@BJC.BGI.BA.BBE
CSeq: 2045 REGISTER
Contact: sip:26@142.183.119.36:59722;reg-id=3;+sip.instance=“urn:uuid:00000000-0000-1000-8000-C074AD3EEF06”
Authorization: Digest username=“26”, realm=“asterisk”, nonce=“14da8f7a”, uri=“sip:184.107.85.181”, response=“3118633cf91830e589dab83d8664ba6f”, algorithm=MD5
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.11.35
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 142.183.119.36:23557 (NAT)
Reliably Transmitting (NAT) to 142.183.119.36:23557:
OPTIONS sip:26@142.183.119.36:59722 SIP/2.0
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK747254e8;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@184.107.85.181;tag=as47948a2d
To: sip:26@142.183.119.36:59722
Contact: sip:asterisk@184.107.85.181:5060
Call-ID: 3704f25c1e3ea6fa65d452c3593d47c1@184.107.85.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.28.0
Date: Wed, 19 Oct 2022 18:20:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to 142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.114:5064;branch=z9hG4bK1474822169;received=142.183.119.36;rport=23557
From: sip:26@184.107.85.181;tag=181936273
To: sip:26@184.107.85.181;tag=as65d43bad
Call-ID: 11532509-5064-1@BJC.BGI.BA.BBE
CSeq: 2045 REGISTER
Server: Asterisk PBX 16.28.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:26@142.183.119.36:59722;expires=3600
Date: Wed, 19 Oct 2022 18:20:57 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘11532509-5064-1@BJC.BGI.BA.BBE’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:142.183.119.36:23557 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 184.107.85.181:5060;branch=z9hG4bK747254e8;rport=5060
From: “asterisk” sip:asterisk@184.107.85.181;tag=as47948a2d
To: sip:26@142.183.119.36:59722;tag=2033396113
Call-ID: 3704f25c1e3ea6fa65d452c3593d47c1@184.107.85.181:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.11.35
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘3704f25c1e3ea6fa65d452c3593d47c1@184.107.85.181:5060’ Method: OPTIONS

sean · October 19, 2022, 6:27pm

[general]
context=public ; Default context for incoming calls. Defaults to ‘default’
allowguest=no ; Allow or reject guest calls (default is yes)
match_auth_username=yes ; if available, match user entry using the
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
; Codec negotiation
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side’s codec choice to exactly what we prefer.

allow=all ; First disallow all codecs
allow=ulaw
allow=alaw
allow=g729 ; see RTP Packetization - Asterisk Project - Asterisk Project Wiki

[25]
type=friend
qualify=yes
call-limit=99
insecure=invite,port
username=25
callerid=25
secret=xx
context=ShabOut
host=dynamic
allow=all

[26]
type=friend
qualify=yes
call-limit=99
insecure=invite,port
username=26
callerid=26
secret=xx
context=ShabOut
host=dynamic
allow=all

when 26 calls 25, 25 can hear, if 25 calls 26 no sound on both ext.

david551 · October 19, 2022, 7:14pm

This should be disallow. Setting too many codecs can break things, although that doesn’t seem to be the problem here.

It fails because you are trying to do direct media between a private and a public address. Specifically, 26 is sending a private address in its SDP (and via header) but actually sending from a public one, and using a a public Contact.

If you can’t fix 26, you will need nat=comedia, for the former. I’m not sure how you handle the bad contact address; I’m not sure if force_rport is sufficient.

Disabling directmedia may also be advisable, as I’m not sure how well it interacts with comedia.

I can’t see a valid reason for type=friend, especially for 25. I haven’t checked closely enough, but insecure=port may actually be needed here, although it is normally incorrectly used.

When posting logs and configuration files, please mark them up as pre-formatted text, otherwise they get garbled by the forum.

Why are you using Asterisk 16? It is in a security fixes only maintenance state, and has been for 10 days, and Asterisk 20 was released a few hours ago. (I believe there are some maintenance fixes in the pipeline for 16, but no further ones will be accepted. Also, why are you using chan_sip. It is deprecated, effectively unsupported, and likely to disappear from the GIT master version any day now.

sean · October 19, 2022, 7:36pm

Thanks David for your time.
I was using asterisk 1.4 for many years and knew how to code sip accounts.
I installed asterisk 18 and couldn’t use sip that’s why i downgraded to 16.
my configuration is this
We have a dedicated server that runs asterisk
right now i have both phones in my office and testing them out.
In the past with asterisk 1.4 we never had an issue calling in the office between extensions.

Is it hard to code config file for accounts for the new res_pjsip? and will our current phones would work with it?
Much appreciate your time.

david551 · October 19, 2022, 7:47pm

I think the default for directmedia may have changed after 1.4.

sean · October 19, 2022, 8:54pm

I did what you told me to do, now even 25 cant hear anything and calls gets dropped after few sec.

david551 · October 19, 2022, 10:22pm

By a few seconds, do you mean about 30? Otherwise I’m not sure what the cause is. In any case, I’d want to see the logging.

My feeling is that the key problem is directmedia defaulting to yes, so you want to set directmedia to no.

The default for nat= is auto for everything and I think it might have triggered the right work arounds. If you replaced that by just nat=comedia, it could have broken nat=force_rport handling. It does seem to have gone into NAT work round mode for 25 “Transmitting (NAT)”, and also for 26, so I think it was defaulting to nat=comedia,force_rport

As noted, it would be much better if you could configure the phone that is sending a mixture of public an private addresses to just send pubic ones

I don’t think that changing type= to peer could break it in that way.

Incidentally, although it is just possible that you need insecure=port, I think insecure=invite is positively dangerous here, especially with type=friend; it means that calls from user agents claiming to be 25 or 26 will be permitted with no authentication. It would be bad practice with type=peer, but the register would have been authenticated and invites only accepted from the IP address that registered, but the type=friend mans that invites will be accepted from anyone with a from user of 25 or 26, as the invite will not be challenged for authentication.

Although allow=all can break things (and hast done so), you should only change it to disallow in the general section, and should remove it completely in for the individual end points, as disallow=all, there will leave you with no possible codecs.

call-limit=99 is weird, but I don’t think it can possibly be reached, so the phones will can appear as in-use, but never busy, in their device states. I think this is another area of change from 1.4, I’m not sure if it has actually being used, but if you are not using device states, it probably has no effect. See:

github.com

asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L117


      
          ; (caller ID) will match any of your device names, because then Asterisk
          ; might match the wrong device.
          ;
          ; Note: The parameter "username" is not the username and in most cases is
          ;       not needed at all. Check below. In later releases, it's renamed
          ;       to "defaultuser" which is a better name, since it is used in
          ;       combination with the "defaultip" setting.
          ; ----------------------------------------------------------------------------
          
          
; ** Old configuration options **
          ; The "call-limit" configuation option is considered old is replaced
          ; by new functionality. To enable callcounters, you use the new
          ; "callcounter" setting (for extension states in queue and subscriptions)
          ; You are encouraged to use the dialplan groupcount functionality
          ; to enforce call limits instead of using this channel-specific method.
          ; You can still set limits per device in sip.conf or in a database by using
          ; "setvar" to set variables that can be used in the dialplan for various limits.
          
          
[general]
          context=public                  ; Default context for incoming calls. Defaults to 'default'
          ;allowguest=no                  ; Allow or reject guest calls (default is yes)

sean · October 20, 2022, 2:29pm

Dave, our old server data center will be closed Oct 27, so I need to have a functioning system before so I can transfer everything.
If you could make some time to have a zoom meeting so we figure this out I would really appreciated. Again I’ll pay for your time.

sean · October 20, 2022, 7:26pm

I installed Astterisk 20 per your advise, and still having issue. Was able to run sip.

Connected to Asterisk 20.0.0 currently running on asterisk2022 (pid = 950)
  == Using SIP RTP CoS mark 5
       > 0x7f1360014a90 -- Strict RTP learning after remote address set to: 142.183.119.36:28308
    -- Executing [25@ShabOut:1] Set("SIP/26-0000000e", "CALLERID(all)="The Visa" <416-477-2545>") in new stack
    -- Executing [25@ShabOut:2] Answer("SIP/26-0000000e", "10") in new stack
    -- Executing [25@ShabOut:3] Dial("SIP/26-0000000e", "SIP/25") in new stack
[Oct 20 19:25:21] NOTICE[1549][C-0000000d]: app_dial.c:2707 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [25@ShabOut:4] Hangup("SIP/26-0000000e", "") in new stack
  == Spawn extension (ShabOut, 25, 4) exited non-zero on 'SIP/26-0000000e'
  == Using SIP RTP CoS mark 5
       > 0x7f13600207f0 -- Strict RTP learning after remote address set to: 142.183.119.36:39212
[Oct 20 19:25:27] WARNING[1356]: chan_sip.c:4153 retrans_pkt: Retransmission timeout reached on transmission 1048049561-5064-29@BJC.BGI.BA.BBE for seqno 280 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

david551 · October 20, 2022, 8:16pm

25 is not registered. Nothing I suggested has any effect on the ability to register.

sean · October 20, 2022, 8:32pm

Here is the log

Connected to Asterisk 20.0.0 currently running on asterisk2022 (pid = 950)
== Using SIP RTP CoS mark 5
> 0x7f136001eb00 – Strict RTP learning after remote address set to: 192.168.10.136:5016
– Executing [26@ShabOut:1] Set(“SIP/25-00000021”, “CALLERID(all)=“The Visa” <416-477-2545>”) in new stack
– Executing [26@ShabOut:2] Answer(“SIP/25-00000021”, “10”) in new stack
– Executing [26@ShabOut:3] Dial(“SIP/25-00000021”, “SIP/26”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/26
– SIP/26-00000022 is ringing
> 0x7f13700065d0 – Strict RTP learning after remote address set to: 142.183.119.36:20382
– SIP/26-00000022 answered SIP/25-00000021
– Channel SIP/26-00000022 joined ‘simple_bridge’ basic-bridge
– Channel SIP/25-00000021 joined ‘simple_bridge’ basic-bridge
> Bridge b3499018-970b-4343-b63c-777b01e85266: switching from simple_bridge technology to native_rtp
> Locally RTP bridged ‘SIP/25-00000021’ and ‘SIP/26-00000022’ in stack
[Oct 20 20:32:37] WARNING[1356]: chan_sip.c:4153 retrans_pkt: Retransmission timeout reached on transmission 873055374-5070-108@BJC.BGI.BA.BDG for seqno 1070 (Critical Response) – See SIP Retransmissions - Asterisk Project - Asterisk Project Wiki
Packet timed out after 6399ms with no response
[Oct 20 20:32:37] WARNING[1356]: chan_sip.c:4177 retrans_pkt: Hanging up call 873055374-5070-108@BJC.BGI.BA.BDG - no reply to our critical packet (see SIP Retransmissions - Asterisk Project - Asterisk Project Wiki).
– Channel SIP/25-00000021 left ‘native_rtp’ basic-bridge
== Spawn extension (ShabOut, 26, 3) exited non-zero on ‘SIP/25-00000021’
– Channel SIP/26-00000022 left ‘native_rtp’ basic-bridge

david551 · October 20, 2022, 8:52pm

It’s not getting an ACK to its OK. You need the SIP trace to fully understand this, but the basic cause is that the device is behind NAT but not handling that properly. Can you confirm that you have set force_rport?

This is probably not getting the OK to the BYE. The above comments. apply.

Given that you say it worked on 1.4, have you tried setting directmedia=no as the only change.

Generally people will be more prepared to work out the correct workaround for chan_pjsip, although it is still best to fix the phone.

sean · October 21, 2022, 4:17pm

hello I trying to move the new pjsip.conf now
I made this:

simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

[35]
type = endpoint
context = ShabOut
allow=ulaw
allow=alaw
allow=g729
allow=gsm
aors = 35
auth = auth35

[35]
type = aor
max_contacts = 1

[auth35]
type=auth
auth_type=userpass
password=##boom
username=35

[36]
type = endpoint
context = ShabOut
allow=ulaw
allow=alaw
allow=g729
allow=gsm
aors = 36
auth = auth36

[36]
type = aor
max_contacts = 1

[auth36]
type=auth
auth_type=userpass
password=##boom
username=36

When i try to register the sip phone i get :

[Oct 21 16:17:29] NOTICE[3259]: chan_sip.c:29060 handle_request_register: Registration from ‘sip:36@34.130.110.224’ failed for ‘142.183.119.36:24010’ - Wrong password

david551 · October 21, 2022, 4:31pm

I can’t see anything that suggests that the password is not wrong.

However, when you get to doing INVITEs, you need to disallow all before allowing codecs.

sean · October 21, 2022, 5:01pm

I changed that, how how do I register phone with it? or soft phone, i tried entered the info but on the server side it shows wrong pass, keep in mind sip.iso is also running

[35]
type = endpoint
context = ShabOut
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
aors = 35
auth = auth35

[35]
type = aor
max_contacts = 1

[auth35]
type=auth
auth_type=userpass
password=##boom
username=35

[36]
type = endpoint
context = ShabOut
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
aors = 36
auth = auth36

[36]
type = aor
max_contacts = 1

[auth36]
type=auth
auth_type=userpass
password=##boom
username=36

david551 · October 21, 2022, 5:03pm

I hadn’t notice that you still had chan_sip, and that was taking the registration. You obviously cannot bind two drivers to the same port.

sean · October 21, 2022, 5:08pm

asterisk2022*CLI> module unload chan_sip.so
Unable to unload resource chan_sip.so
Command ‘module unload chan_sip.so’ failed.
[Oct 21 17:08:05] WARNING[3511]: loader.c:1239 ast_unload_resource: Unload failed, ‘chan_sip.so’ could not be found

i unload it that

Asterisk 16 brand new install cant not hear sound between 2 calls

[26] type=friend qualify=yes call-limit=99 insecure=invite,port username=26 callerid=26 secret=xx context=ShabOut host=dynamic allow=all

[26]
type=friend
qualify=yes
call-limit=99
insecure=invite,port
username=26
callerid=26
secret=xx
context=ShabOut
host=dynamic
allow=all