Problem with Asterisk AMI after switching from chan_sip (Asterisk 18.8.0) to pjsip (Asterisk 22.7.0)

Asterisk Asterisk APIs
1

In my C++ application I use the Asterisk AMI to get information about registered peers. For this I use the AMI “PeerStatus” event message.

Configuration of chan_sip/sip.conf:

[general]
bindport=5060,5061
disallow=all
allow=g722
allow=alaw
minexpiry=5
defaultexpiry=15
maxexpiry=15

[74500]
type=peer
host=dynamic
transport=udp
context=extern
dtmfmode=info
directmedia=off
disallow=all
allow=alaw
allow=ulaw
qualify=2000
qualifyfreq=10

With this configuration I get a cyclic AMI event message “PeerStatus” every 15 seconds.

For migration from chan_sip to pjsip I used the Python script sip_to_pjsip.py:

[74500]
type = endpoint
context = extern
dtmf_mode = info
disallow = all
allow = alaw
allow = ulaw
rtp_timeout = 10
sdp_owner = dxc
sdp_session = dxc
aors = 74500

[74500]
type = aor
max_contacts = 1
qualify_frequency = 10
maximum_expiration = 15
minimum_expiration = 5
default_expiration = 15

But with this pjsip configuration the related contact will be deleted every 15 seconds and peer state toggles from available to unavailable and then availabe againg:

== Contact 74500/sip:74500@10.60.10.164:5060 has been deleted
== Endpoint 74500 is now Unreachable
– Added contact ‘sip:74500@10.60.10.164:5060’ to AOR ‘74500’ with expiration of 15 seconds
← Examining AMI event (2064270355): →
Event: PeerStatus
Privilege: system,all
SequenceNumber: 16
File: manager.c
Line: 597
Func: manager_default_msg_cb
ChannelType: PJSIP
Peer: PJSIP/74500
PeerStatus: Unreachable

← Examining AMI event (541258673): →
Event: ContactStatus
Privilege: system,all
SequenceNumber: 17
File: manager.c
Line: 597
Func: manager_default_msg_cb
URI: sip:74500@10.60.10.164:5060
ContactStatus: Removed
AOR: 74500
EndpointName: 74500
RoundtripUsec: 0

← Examining AMI event (541258673): →
Event: ContactStatus
Privilege: system,all
SequenceNumber: 18
File: manager.c
Line: 597
Func: manager_default_msg_cb
URI: sip:74500@10.60.10.164:5060
ContactStatus: NonQualified
AOR: 74500
EndpointName: 74500
RoundtripUsec: 0

== Endpoint 74500 is now Reachable
– Contact 74500/sip:74500@10.60.10.164:5060 is now Reachable. RTT: 319.311 msec
← Examining AMI event (2064270355): →
Event: PeerStatus
Privilege: system,all
SequenceNumber: 19
File: manager.c
Line: 597
Func: manager_default_msg_cb
ChannelType: PJSIP
Peer: PJSIP/74500
PeerStatus: Reachable

← Examining AMI event (541258673): →
Event: ContactStatus
Privilege: system,all
SequenceNumber: 20
File: manager.c
Line: 597
Func: manager_default_msg_cb
URI: sip:74500@10.60.10.164:5060
ContactStatus: Reachable
AOR: 74500
EndpointName: 74500
RoundtripUsec: 319311

localhost*CLI>

How can I aviod this problem so that the peer state keeps available as long as the peer device is connected?

Any help is appreciated.

2

This would seem to indicate it’s not connected though. Is it actually re-registering before expiration?

3

Device is connected :wink:

And device’s registering timeout is set to 60 minutes.

Although when using chan_sip every 15 Seconds I get a pair of this:

← Examining AMI event: →
Event: PeerStatus
Privilege: system,all
SequenceNumber: 2130
File: manager.c
Line: 1863
Func: manager_default_msg_cb
ChannelType: SIP
Peer: SIP/75500
PeerStatus: Registered
Address: 10.60.10.164:5064

← Examining AMI event: →
Event: SuccessfulAuth
Privilege: security,all
SequenceNumber: 2131
File: manager.c
Line: 1863
Func: manager_default_msg_cb
EventTV: 2026-01-23T10:41:03.781+0000
Severity: Informational
Service: SIP
EventVersion: 1
AccountID: 75500
SessionID: 0xb7313188
LocalAddress: IPV4/UDP/10.60.106.2/5060
RemoteAddress: IPV4/UDP/10.60.10.164/5064
UsingPassword: 0

4

You’ve configured the server to enforce the registration time to be every 15 seconds, so even if the client is set for 60 minutes it wouldn’t be used. 15 seconds is extremely aggressive. The client has to re-register before then, if it doesn’t then it’ll be deleted and go unreachable. As you haven’t provided timestamped SIP traffic, I can’t say whether it is doing that or not.

5

localhost*CLI> pjsip show history
No. Timestamp (Dir) Address SIP Message
===== ========== ==================== ===================================
00000 1769769549 * <== 10.60.10.164:5060 REGISTER sip:10.60.106.1 SIP/2.0
00001 1769769549 * ==> 10.60.10.164:5060 SIP/2.0 200 OK
00002 1769769549 * ==> 10.60.10.164:5060 OPTIONS sip:74500@10.60.10.164:5060 SIP/2.0
00003 1769769550 * <== 10.60.10.164:5060 SIP/2.0 200 OK
00004 1769769566 * <== 10.60.10.164:5060 REGISTER sip:10.60.106.1 SIP/2.0
00005 1769769566 * ==> 10.60.10.164:5060 SIP/2.0 200 OK
00006 1769769566 * ==> 10.60.10.164:5060 OPTIONS sip:74500@10.60.10.164:5060 SIP/2.0
00007 1769769566 * <== 10.60.10.164:5060 SIP/2.0 200 OK

Every time the register message arrives the related contact will be removed and new created immediately:

<— Received SIP request (580 bytes) from UDP:10.60.10.164:5060 —>
REGISTER sip:10.60.106.1 SIP/2.0
Via: SIP/2.0/UDP 10.60.10.164:5060;branch=z9hG4bK1748439571;rport
From: sip:74500@10.60.106.1;tag=1627835864
To: sip:74500@10.60.106.1
Call-ID: 527315809-5060-1@BA.GA.BA.BGE
CSeq: 2267 REGISTER
Contact: sip:74500@10.60.10.164:5060;reg-id=1;+sip.instance=“urn:uuid:00000000-0000-1000-8000-000B82ABDBC8”
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP1782 1.0.1.126
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

== Contact 74500/sip:74500@10.60.10.164:5060 has been deleted
== Endpoint 74500 is now Unreachable
– Added contact ‘sip:74500@10.60.10.164:5060’ to AOR ‘74500’ with expiration of 15 seconds
<— Transmitting SIP response (404 bytes) to UDP:10.60.10.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.10.164:5060;rport=5060;received=10.60.10.164;branch=z9hG4bK1748439571
Call-ID: 527315809-5060-1@BA.GA.BA.BGE
From: sip:74500@10.60.106.1;tag=1627835864
To: sip:74500@10.60.106.1;tag=z9hG4bK1748439571
CSeq: 2267 REGISTER
Date: Fri, 30 Jan 2026 10:45:55 GMT
Contact: sip:74500@10.60.10.164:5060;expires=14
Expires: 15
Server: dxc
Content-Length: 0

<— Transmitting SIP request (405 bytes) to UDP:10.60.10.164:5060 —>
OPTIONS sip:74500@10.60.10.164:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj7b8682e4-47ef-4960-8d7f-e617cc598dd5
From: sip:74500@10.60.106.1;tag=d954d0a2-f133-4c5d-9801-ef0ce23e0551
To: sip:74500@10.60.10.164
Contact: sip:74500@10.60.106.1:5060
Call-ID: c92057f4-fb8c-4a6d-a6e2-3f3cd9159a15
CSeq: 54631 OPTIONS
Max-Forwards: 70
User-Agent: dxc
Content-Length: 0

<— Received SIP response (488 bytes) from UDP:10.60.10.164:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj7b8682e4-47ef-4960-8d7f-e617cc598dd5
From: sip:74500@10.60.106.1;tag=d954d0a2-f133-4c5d-9801-ef0ce23e0551
To: sip:74500@10.60.10.164;tag=876181589
Call-ID: c92057f4-fb8c-4a6d-a6e2-3f3cd9159a15
CSeq: 54631 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1782 1.0.1.126
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

== Endpoint 74500 is now Reachable
– Contact 74500/sip:74500@10.60.10.164:5060 is now Reachable. RTT: 322.003 msec

6

You have configured a maximum expiration of 15. According to your history, the remote side is NOT re-registering before expiration. In the given history it registered after 17 seconds. This would result in the Contact being removed due to expiration at 15 seconds, and then shortly after it would be registered again. I would suggest increasing the maximum expiration to something like 60 seconds.

7

I am one step further: with below settings the endpoint is no longer deleted:

[74500]
type = aor
max_contacts = 1
maximum_expiration = 20
minimum_expiration = 10
default_expiration = 10
qualify_frequency = 15
remove_existing = yes

As my application monitors the cyclic received register messages I now have a different problem: Is there any possibilty to get an AMI event vor every received register message (not only for the first one)? Or do I have to actively query the status (using which command?)?

8

There is a global option that can be enabled which will cause ContactStatus to be sent for registration refreshes:

9

Yes - I found it :wink:

Thank you very much for all your help!

10

I don’t think the designers of SIP would have expected such short timeouts!

11

@david551 Not just SIP, but also SIP implementations…

12

Yes I know, but the SIP clients are here used in a plant warning system with continuous monitoring and automatic fault notication. So there are short timeouts required.

13

Now I try to setup first call :slightly_smiling_face: - but nothing works so far.

From my sip.conf:

[dxc]
disallow=all
allow=g722
allow=alaw
transport=udp
qualify=no
qualifyfreq=0
minexpiry=0
defaultexpiry=0
maxexpiry=0
rtptimeout=0
context=intern
type=peer
insecure=invite,port
bindaddr=127.0.0.1:33099
host=127.0.0.1
dtmfmode=info
directmedia=update

The Python script sip_to_pjsip.py has not used the configuration of “bindaddr=127.0.0.1:33099”:

pjsip.conf:

[dxc]
type = aor
;contact = sip:127.0.0.1:33099 this also doesn’t work
contact = sip:127.0.0.1
maximum_expiration = 0
minimum_expiration = 0
default_expiration = 0
qualify_frequency = 0

[dxc]
type = identify
endpoint = dxc
match = 127.0.0.1

[dxc]
type = endpoint
context = intern
transport = transport-udp
dtmf_mode = info
rtp_timeout = 0
sdp_owner = dxc
sdp_session = dxc
aors = dxc
disallow = all
allow = g722
allow = alaw
direct_media_method = update

This is my extension.conf (I only replaced SIP by PJSIP):

[intern]
exten => 74500,1,GotoIf($[x${SIPPEER(74500,status):0:2} = xOK]?2:3)
exten => 74500,2,Dial(PJSIP/${EXTEN}@74500,30)
exten => 74500,3,Hangup()
exten => 74501,1,GotoIf($[x${SIPPEER(74501,status):0:2} = xOK]?2:3)
exten => 74501,2,Dial(PJSIP/${EXTEN}@74501,30)
exten => 74501,3,Hangup()
exten => 74601,1,GotoIf($[x${SIPPEER(74601,status):0:2} = xOK]?2:3)
exten => 74601,2,Dial(PJSIP/${EXTEN}@74601,30)
exten => 74601,3,Hangup()
exten => 920,1,GotoIf($[x${SIPPEER(920,status):0:2} = xOK]?2:3)
exten => 920,2,Dial(PJSIP/${EXTEN}@920,30)
exten => 920,3,Hangup()
exten => 74600,1,GotoIf($[x${SIPPEER(74600,status):0:2} = xOK]?2:3)
exten => 74600,2,Dial(PJSIP/${EXTEN}@74600,30)
exten => 74600,3,Hangup()
exten => 1000,1,Dial(PJSIP/${EXTEN}@DXC75,30)
exten => 75500,1,Dial(PJSIP/${EXTEN}@DXC75,30)
exten => 75501,1,Dial(PJSIP/${EXTEN}@DXC75,30)
exten => _X.,1,Set(CALLERID(name)=)
exten => _X.,2,GotoIf($[x${SIPPEER(888,status):0:2} = xOK]?3:4)
exten => _X.,3,Dial(PJSIP/${EXTEN}@888,30)
exten => _X.,4,Hangup()
[extern]
exten => _X.,1,Dial(PJSIP/${EXTEN}@localhost:33099)
exten => _+X.,1,Goto(extern,00${EXTEN:1},1)

Asterisk Log:

[Feb 3 09:10:15] <— Received SIP request (1202 bytes) from UDP:10.60.10.164:5060 —>
[Feb 3 09:10:15] INVITE sip:7450174501@10.60.106.1 SIP/2.0
[Feb 3 09:10:15] Via: SIP/2.0/UDP 10.60.10.164:5060;branch=z9hG4bK366207613;rport
[Feb 3 09:10:15] From: “74500” sip:74500@10.60.106.1;tag=183154812
[Feb 3 09:10:15] To: sip:7450174501@10.60.106.1
[Feb 3 09:10:15] Call-ID: 375302356-5060-10@BA.GA.BA.BGE
[Feb 3 09:10:15] CSeq: 90 INVITE
[Feb 3 09:10:15] Contact: “74500” sip:74500@10.60.10.164:5060
[Feb 3 09:10:15] X-Grandstream-PBX: true
[Feb 3 09:10:15] Max-Forwards: 70
[Feb 3 09:10:15] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 3 09:10:15] Privacy: none
[Feb 3 09:10:15] P-Preferred-Identity: “74500” sip:74500@10.60.106.1
[Feb 3 09:10:15] P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-1B-17-00-01-40
[Feb 3 09:10:15] P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-AB-DB-C8
[Feb 3 09:10:15] Supported: replaces, path, timer
[Feb 3 09:10:15] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 3 09:10:15] Content-Type: application/sdp
[Feb 3 09:10:15] Accept: application/sdp, application/dtmf-relay
[Feb 3 09:10:15] Content-Length: 387
[Feb 3 09:10:15]
[Feb 3 09:10:15] v=0
[Feb 3 09:10:15] o=74500 8000 8000 IN IP4 10.60.10.164
[Feb 3 09:10:15] s=SIP Call
[Feb 3 09:10:15] c=IN IP4 10.60.10.164
[Feb 3 09:10:15] t=0 0
[Feb 3 09:10:15] m=audio 5004 RTP/AVP 8 0 4 9 123 97 2 101
[Feb 3 09:10:15] a=sendrecv
[Feb 3 09:10:15] a=rtpmap:8 PCMA/8000
[Feb 3 09:10:15] a=ptime:20
[Feb 3 09:10:15] a=rtpmap:0 PCMU/8000
[Feb 3 09:10:15] a=rtpmap:4 G723/8000
[Feb 3 09:10:15] a=rtpmap:9 G722/8000
[Feb 3 09:10:15] a=rtpmap:123 opus/48000/2
[Feb 3 09:10:15] a=rtpmap:97 iLBC/8000
[Feb 3 09:10:15] a=fmtp:97 mode=30
[Feb 3 09:10:15] a=rtpmap:2 G726-32/8000
[Feb 3 09:10:15] a=rtpmap:101 telephone-event/8000
[Feb 3 09:10:15] a=fmtp:101 0-15
[Feb 3 09:10:15]
[Feb 3 09:10:15] <— Transmitting SIP response (293 bytes) to UDP:10.60.10.164:5060 —>
[Feb 3 09:10:15] SIP/2.0 100 Trying
[Feb 3 09:10:15] Via: SIP/2.0/UDP 10.60.10.164:5060;rport=5060;received=10.60.10.164;branch=z9hG4bK366207613
[Feb 3 09:10:15] Call-ID: 375302356-5060-10@BA.GA.BA.BGE
[Feb 3 09:10:15] From: “74500” sip:74500@10.60.106.1;tag=183154812
[Feb 3 09:10:15] To: sip:7450174501@10.60.106.1
[Feb 3 09:10:15] CSeq: 90 INVITE
[Feb 3 09:10:15] Server: dxc
[Feb 3 09:10:15] Content-Length: 0
[Feb 3 09:10:15]
[Feb 3 09:10:15]
[Feb 3 09:10:15] – Executing [7450174501@extern:1] Dial(“PJSIP/74500-00000003”, “PJSIP/7450174501@localhost:33099”) in new stack
[Feb 3 09:10:15] ERROR[9321]: chan_pjsip.c:2715 request: Unable to create PJSIP channel - endpoint ‘localhost:33099’ was not found
[Feb 3 09:10:15] NOTICE[10324][C-00000004]: app_dial.c:2722 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
[Feb 3 09:10:15] == Everyone is busy/congested at this time (1:0/0/1)
[Feb 3 09:10:15] – Auto fallthrough, channel ‘PJSIP/74500-00000003’ status is ‘CHANUNAVAIL’
[Feb 3 09:10:15] <— Transmitting SIP response (371 bytes) to UDP:10.60.10.164:5060 —>
[Feb 3 09:10:15] SIP/2.0 503 Service Unavailable
[Feb 3 09:10:15] Via: SIP/2.0/UDP 10.60.10.164:5060;rport=5060;received=10.60.10.164;branch=z9hG4bK366207613
[Feb 3 09:10:15] Call-ID: 375302356-5060-10@BA.GA.BA.BGE
[Feb 3 09:10:15] From: “74500” sip:74500@10.60.106.1;tag=183154812
[Feb 3 09:10:15] To: sip:7450174501@10.60.106.1;tag=c66f2556-5913-4fc5-8401-b39df8588548
[Feb 3 09:10:15] CSeq: 90 INVITE
[Feb 3 09:10:15] Server: dxc
[Feb 3 09:10:15] Reason: Q.850;cause=34
[Feb 3 09:10:15] Content-Length: 0
[Feb 3 09:10:15]
[Feb 3 09:10:15]
[Feb 3 09:10:16] <— Received SIP request (310 bytes) from UDP:10.60.10.164:5060 —>
[Feb 3 09:10:16] ACK sip:7450174501@10.60.106.1 SIP/2.0
[Feb 3 09:10:16] Via: SIP/2.0/UDP 10.60.10.164:5060;branch=z9hG4bK366207613;rport
[Feb 3 09:10:16] From: “74500” sip:74500@10.60.106.1;tag=183154812
[Feb 3 09:10:16] To: sip:7450174501@10.60.106.1;tag=c66f2556-5913-4fc5-8401-b39df8588548
[Feb 3 09:10:16] Call-ID: 375302356-5060-10@BA.GA.BA.BGE
[Feb 3 09:10:16] CSeq: 90 ACK
[Feb 3 09:10:16] Content-Length: 0

sip:7450174501@10.60.106.1 looks strange :roll_eyes:

What do I miss?

14

This is not valid.

If you want to dial a SIP URI, you need to specify an endpoint with it. Such as:

PJSIP/dxc/sip:${EXTEN}@localhost:33099

15

Next Problem:
When Direct Media is set to “Udate” then whenever I accept a call at the destination SIP phone the call will be hung up. This doesn’t happen, if Direct Media is not used at all.
Any ideas?

[Feb 6 09:28:08] – Executing [74500@intern:1] GotoIf(“PJSIP/dxc-0000000a”, “1?2:3”) in new stack
[Feb 6 09:28:08] – Goto (intern,74500,2)
[Feb 6 09:28:08] – Executing [74500@intern:2] Dial(“PJSIP/dxc-0000000a”, “PJSIP/74500@74500,30”) in new stack
[Feb 6 09:28:08] – Called PJSIP/74500@74500
[Feb 6 09:28:08] – PJSIP/74500-0000000b is ringing

Call accepted:
[Feb 6 09:28:15] > 0xb4403b20 – Strict RTP learning after remote address set to: 10.60.10.164:5004
[Feb 6 09:28:15] – PJSIP/74500-0000000b answered PJSIP/dxc-0000000a
[Feb 6 09:28:15] > 0xb6f9bef0 – Strict RTP learning after remote address set to: 10.60.106.1:35010
[Feb 6 09:28:15] – Channel PJSIP/74500-0000000b joined ‘simple_bridge’ basic-bridge
[Feb 6 09:28:15] – Channel PJSIP/dxc-0000000a joined ‘simple_bridge’ basic-bridge
[Feb 6 09:28:15] > Bridge b12a890f-ebbd-4434-ae51-591fe9c51d85: switching from simple_bridge technology to native_rtp
[Feb 6 09:28:15] > Remotely bridged ‘PJSIP/dxc-0000000a’ and ‘PJSIP/74500-0000000b’ - media will flow directly between them
[Feb 6 09:28:15] – Channel PJSIP/dxc-0000000a left ‘native_rtp’ basic-bridge
[Feb 6 09:28:15] == Spawn extension (intern, 74500, 2) exited non-zero on ‘PJSIP/dxc-0000000a’
[Feb 6 09:28:15] – Channel PJSIP/74500-0000000b left ‘native_rtp’ basic-bridge

From a call before with more debug info:
[Feb 6 09:06:21] VERBOSE[8192][C-00000005] bridge_channel.c: Channel PJSIP/dxc-00000008 left ‘native_rtp’ basic-bridge
[Feb 6 09:06:21] VERBOSE[8192][C-00000005] pbx.c: Spawn extension (intern, 74500, 2) exited non-zero on ‘PJSIP/dxc-00000008’
[Feb 6 09:06:21] VERBOSE[8004] manager.c: ← Examining AMI event (1679088753): →
Event: HangupRequest
Privilege: call,all
SequenceNumber: 226
File: manager_channels.c
Line: 873
Func: channel_hangup_request_cb
Channel: PJSIP/dxc-00000008
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 74005
CallerIDName: Digital Station 74005
ConnectedLineNum: 74500
ConnectedLineName:
Language: en
AccountCode:
Context: intern
Exten: 74500
Priority: 2
Uniqueid: 1770368770.8
Linkedid: 1770368770.8
Cause: 16
TechCause: 400

I have to mention that the initial INVITE message is generated by separate software application using a libosip interface to asterisk.

16

A SIP trace is required to see what actually happened.

17

Is a Wireshark screenshot sufficient or something else?

18

The output of “pjsip set logger on” generally is best. For a screenshot from Wireshark you would need multiple to show the contents of the complete SIP interaction with complete SIP packets.

19

Direct Media is enabled and set to Update.

Connected to Asterisk 22.7.0 currently running on localhost (pid = 7956)
[Feb 6 09:47:58] <— Received SIP request (695 bytes) from UDP:127.0.0.1:33099 —>
[Feb 6 09:47:58] INVITE sip:74500@127.0.0.1:5060;user=phone SIP/2.0
[Feb 6 09:47:58] Via: SIP/2.0/UDP 127.0.0.1:33099;branch=z9hG4bK1957436329
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@127.0.0.1:33099;tag=1876607352
[Feb 6 09:47:58] To: sip:74500@127.0.0.1:5060
[Feb 6 09:47:58] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:47:58] CSeq: 102 INVITE
[Feb 6 09:47:58] Contact: sip:74005@127.0.0.1:33099;transport=udp;user=phone
[Feb 6 09:47:58] Content-Type: application/sdp
[Feb 6 09:47:58] Allow: INVITE, ACK, BYE, CANCEL, INFO, NOTIFY, OPTIONS, PRACK, REFER, UPDATE, MESSAGE
[Feb 6 09:47:58] Content-Length: 217
[Feb 6 09:47:58]
[Feb 6 09:47:58] v=0
[Feb 6 09:47:58] o=74005 1770371278 1770371278 IN IP4 10.60.106.1
[Feb 6 09:47:58] s=A conversation
[Feb 6 09:47:58] c=IN IP4 10.60.106.1
[Feb 6 09:47:58] t=0 0
[Feb 6 09:47:58] m=audio 35016 RTP/AVP 8 101
[Feb 6 09:47:58] a=rtpmap:8 PCMA/8000
[Feb 6 09:47:58] a=rtpmap:101 telephone-event/8000
[Feb 6 09:47:58] a=fmtp:101 0-11
[Feb 6 09:47:58] a=sendrecv
[Feb 6 09:47:58]
[Feb 6 09:47:58] <— Transmitting SIP response (295 bytes) to UDP:127.0.0.1:33099 —>
[Feb 6 09:47:58] SIP/2.0 100 Trying
[Feb 6 09:47:58] Via: SIP/2.0/UDP 127.0.0.1:33099;rport=33099;received=127.0.0.1;branch=z9hG4bK1957436329
[Feb 6 09:47:58] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:47:58] To: sip:74500@127.0.0.1
[Feb 6 09:47:58] CSeq: 102 INVITE
[Feb 6 09:47:58] Server: dxc
[Feb 6 09:47:58] Content-Length: 0
[Feb 6 09:47:58]
[Feb 6 09:47:58]
[Feb 6 09:47:58] – Executing [74500@intern:1] GotoIf(“PJSIP/dxc-00000010”, “1?2:3”) in new stack
[Feb 6 09:47:58] – Goto (intern,74500,2)
[Feb 6 09:47:58] – Executing [74500@intern:2] Dial(“PJSIP/dxc-00000010”, “PJSIP/74500@74500,30”) in new stack
[Feb 6 09:47:58] <— Transmitting SIP request (873 bytes) to UDP:10.60.10.164:5060 —>
[Feb 6 09:47:58] INVITE sip:74500@10.60.10.164:5060 SIP/2.0
[Feb 6 09:47:58] Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj75c80f76-3238-4d7a-9ad8-f77f598a4f69
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:47:58] To: sip:74500@10.60.10.164
[Feb 6 09:47:58] Contact: sip:asterisk@10.60.106.1:5060
[Feb 6 09:47:58] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:47:58] CSeq: 8976 INVITE
[Feb 6 09:47:58] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
[Feb 6 09:47:58] Supported: 100rel, timer, replaces, norefersub, histinfo
[Feb 6 09:47:58] Session-Expires: 1800
[Feb 6 09:47:58] Min-SE: 90
[Feb 6 09:47:58] Max-Forwards: 70
[Feb 6 09:47:58] User-Agent: dxc
[Feb 6 09:47:58] Content-Type: application/sdp
[Feb 6 09:47:58] Content-Length: 200
[Feb 6 09:47:58]
[Feb 6 09:47:58] v=0
[Feb 6 09:47:58] o=dxc 1720138998 1720138998 IN IP4 10.60.106.1
[Feb 6 09:47:58] s=dxc
[Feb 6 09:47:58] c=IN IP4 10.60.106.1
[Feb 6 09:47:58] t=0 0
[Feb 6 09:47:58] m=audio 29860 RTP/AVP 8 0
[Feb 6 09:47:58] a=rtpmap:8 PCMA/8000
[Feb 6 09:47:58] a=rtpmap:0 PCMU/8000
[Feb 6 09:47:58] a=ptime:20
[Feb 6 09:47:58] a=maxptime:140
[Feb 6 09:47:58] a=sendrecv
[Feb 6 09:47:58]
[Feb 6 09:47:58] – Called PJSIP/74500@74500
[Feb 6 09:47:58] <— Received SIP response (500 bytes) from UDP:10.60.10.164:5060 —>
[Feb 6 09:47:58] SIP/2.0 100 Trying
[Feb 6 09:47:58] Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj75c80f76-3238-4d7a-9ad8-f77f598a4f69
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:47:58] To: sip:74500@10.60.10.164
[Feb 6 09:47:58] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:47:58] CSeq: 8976 INVITE
[Feb 6 09:47:58] Supported: replaces, path, timer
[Feb 6 09:47:58] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 6 09:47:58] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 6 09:47:58] Content-Length: 0
[Feb 6 09:47:58]
[Feb 6 09:47:58]
[Feb 6 09:47:58] <— Received SIP response (582 bytes) from UDP:10.60.10.164:5060 —>
[Feb 6 09:47:58] SIP/2.0 180 Ringing
[Feb 6 09:47:58] Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj75c80f76-3238-4d7a-9ad8-f77f598a4f69
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:47:58] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:47:58] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:47:58] CSeq: 8976 INVITE
[Feb 6 09:47:58] Contact: sip:74500@10.60.10.164:5060
[Feb 6 09:47:58] Supported: replaces, path, timer
[Feb 6 09:47:58] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 6 09:47:58] Allow-Events: talk, hold
[Feb 6 09:47:58] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 6 09:47:58] Content-Length: 0
[Feb 6 09:47:58]
[Feb 6 09:47:58]
[Feb 6 09:47:58] – PJSIP/74500-00000011 is ringing
[Feb 6 09:47:58] <— Transmitting SIP response (485 bytes) to UDP:127.0.0.1:33099 —>
[Feb 6 09:47:58] SIP/2.0 180 Ringing
[Feb 6 09:47:58] Via: SIP/2.0/UDP 127.0.0.1:33099;rport=33099;received=127.0.0.1;branch=z9hG4bK1957436329
[Feb 6 09:47:58] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:47:58] From: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:47:58] To: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:47:58] CSeq: 102 INVITE
[Feb 6 09:47:58] Server: dxc
[Feb 6 09:47:58] Contact: sip:127.0.0.1:5060
[Feb 6 09:47:58] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
[Feb 6 09:47:58] Content-Length: 0
[Feb 6 09:47:58]
[Feb 6 09:47:58]
[Feb 6 09:48:00] <— Received SIP response (819 bytes) from UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] SIP/2.0 200 OK
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj75c80f76-3238-4d7a-9ad8-f77f598a4f69
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8976 INVITE
[Feb 6 09:48:00] Contact: sip:74500@10.60.10.164:5060
[Feb 6 09:48:00] Supported: replaces, path, timer
[Feb 6 09:48:00] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 6 09:48:00] Session-Expires: 1800;refresher=uac
[Feb 6 09:48:00] Require: timer
[Feb 6 09:48:00] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 180
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=74500 8000 8000 IN IP4 10.60.10.164
[Feb 6 09:48:00] s=SIP Call
[Feb 6 09:48:00] c=IN IP4 10.60.10.164
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 5004 RTP/AVP 8 0
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=rtpmap:0 PCMU/8000
[Feb 6 09:48:00]
[Feb 6 09:48:00] > 0xb6fa0c40 – Strict RTP learning after remote address set to: 10.60.10.164:5004
[Feb 6 09:48:00] <— Transmitting SIP request (396 bytes) to UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] ACK sip:74500@10.60.10.164:5060 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj84631156-e747-4a23-834f-abe926579aa5
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8976 ACK
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] – PJSIP/74500-00000011 answered PJSIP/dxc-00000010
[Feb 6 09:48:00] > 0xb4403b20 – Strict RTP learning after remote address set to: 10.60.106.1:35016
[Feb 6 09:48:00] <— Transmitting SIP response (734 bytes) to UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] SIP/2.0 200 OK
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:33099;rport=33099;received=127.0.0.1;branch=z9hG4bK1957436329
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:48:00] To: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] CSeq: 102 INVITE
[Feb 6 09:48:00] Server: dxc
[Feb 6 09:48:00] Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
[Feb 6 09:48:00] Contact: sip:127.0.0.1:5060
[Feb 6 09:48:00] Supported: 100rel, timer, replaces, norefersub
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 172
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=dxc 1770371278 1770371280 IN IP4 127.0.0.1
[Feb 6 09:48:00] s=dxc
[Feb 6 09:48:00] c=IN IP4 127.0.0.1
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 29360 RTP/AVP 8
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=maxptime:140
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00]
[Feb 6 09:48:00] – Channel PJSIP/74500-00000011 joined ‘simple_bridge’ basic-bridge <341fd643-d78c-42ff-b9b6-fba2140dedca>
[Feb 6 09:48:00] – Channel PJSIP/dxc-00000010 joined ‘simple_bridge’ basic-bridge <341fd643-d78c-42ff-b9b6-fba2140dedca>
[Feb 6 09:48:00] > Bridge 341fd643-d78c-42ff-b9b6-fba2140dedca: switching from simple_bridge technology to native_rtp
[Feb 6 09:48:00] > Remotely bridged ‘PJSIP/dxc-00000010’ and ‘PJSIP/74500-00000011’ - media will flow directly between them
[Feb 6 09:48:00] <— Transmitting SIP request (707 bytes) to UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] UPDATE sip:74005@127.0.0.1:33099 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPjd4f9a870-686b-4e00-8279-b3757716e615
[Feb 6 09:48:00] From: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] To: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:48:00] Contact: sip:127.0.0.1:5060
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] CSeq: 1188 UPDATE
[Feb 6 09:48:00] Supported: 100rel, timer, replaces, norefersub
[Feb 6 09:48:00] Session-Expires: 1800
[Feb 6 09:48:00] Min-SE: 90
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 176
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=dxc 1770371278 1770371281 IN IP4 10.60.106.1
[Feb 6 09:48:00] s=dxc
[Feb 6 09:48:00] c=IN IP4 10.60.10.164
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 5004 RTP/AVP 8
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=maxptime:140
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Transmitting SIP request (751 bytes) to UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] UPDATE sip:74500@10.60.10.164:5060 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj6a9d0d08-635a-42a9-8c79-224c9dc0c856
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Contact: sip:asterisk@10.60.106.1:5060
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8977 UPDATE
[Feb 6 09:48:00] Supported: 100rel, timer, replaces, norefersub
[Feb 6 09:48:00] Session-Expires: 1800;refresher=uac
[Feb 6 09:48:00] Min-SE: 90
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 176
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=dxc 1720138998 1720138999 IN IP4 10.60.106.1
[Feb 6 09:48:00] s=dxc
[Feb 6 09:48:00] c=IN IP4 10.60.106.1
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 35016 RTP/AVP 8
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=maxptime:140
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Received SIP request (422 bytes) from UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] ACK sip:74500@127.0.0.1:5060;user=phone SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:33099;branch=z9hG4bK1957436329
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@127.0.0.1:33099;tag=1876607352
[Feb 6 09:48:00] To: sip:74500@127.0.0.1:5060;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] CSeq: 102 ACK
[Feb 6 09:48:00] Contact: sip:74005@127.0.0.1:33099;transport=udp;user=phone
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Transmitting SIP request (401 bytes) to UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] BYE sip:74005@127.0.0.1:33099 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPjb5559cf1-e737-4d90-a0ad-fa4c1b92644b
[Feb 6 09:48:00] From: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] To: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] CSeq: 1189 BYE
[Feb 6 09:48:00] Reason: Q.850;cause=16
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Received SIP response (587 bytes) from UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] SIP/2.0 200 OK
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPjd4f9a870-686b-4e00-8279-b3757716e615
[Feb 6 09:48:00] From: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] To: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] CSeq: 1188 UPDATE
[Feb 6 09:48:00] Contact: sip:74005@localhost:33099
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] User-Agent: DXC
[Feb 6 09:48:00] Content-Length: 176
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=dxc 1770371278 1770371281 IN IP4 10.60.106.1
[Feb 6 09:48:00] s=dxc
[Feb 6 09:48:00] c=IN IP4 10.60.10.164
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 5004 RTP/AVP 8
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=maxptime:140
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00]
[Feb 6 09:48:00] > 0xb4403b20 – Strict RTP learning after remote address set to: 10.60.10.164:5004
[Feb 6 09:48:00] <— Received SIP response (373 bytes) from UDP:127.0.0.1:33099 —>
[Feb 6 09:48:00] SIP/2.0 200 OK
[Feb 6 09:48:00] Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPjb5559cf1-e737-4d90-a0ad-fa4c1b92644b
[Feb 6 09:48:00] From: sip:74500@127.0.0.1;tag=58948a3f-4060-4ea2-88b6-9c38aebdeece
[Feb 6 09:48:00] To: “Digital Station 74005” sip:74005@127.0.0.1;tag=1876607352
[Feb 6 09:48:00] Call-ID: 178431233-SIPTERM(74.64-4)
[Feb 6 09:48:00] CSeq: 1189 BYE
[Feb 6 09:48:00] Contact: sip:74005@localhost:33099
[Feb 6 09:48:00] User-Agent: DXC
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] – Channel PJSIP/dxc-00000010 left ‘native_rtp’ basic-bridge <341fd643-d78c-42ff-b9b6-fba2140dedca>
[Feb 6 09:48:00] == Spawn extension (intern, 74500, 2) exited non-zero on ‘PJSIP/dxc-00000010’
[Feb 6 09:48:00] – Channel PJSIP/74500-00000011 left ‘native_rtp’ basic-bridge <341fd643-d78c-42ff-b9b6-fba2140dedca>
[Feb 6 09:48:00] <— Transmitting SIP request (420 bytes) to UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] BYE sip:74500@10.60.10.164:5060 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj77f04a58-0641-4caa-a9c4-4becffc67fc9
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8978 BYE
[Feb 6 09:48:00] Reason: Q.850;cause=16
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Received SIP response (548 bytes) from UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] SIP/2.0 200 OK
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj77f04a58-0641-4caa-a9c4-4becffc67fc9
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8978 BYE
[Feb 6 09:48:00] Contact: sip:74500@10.60.10.164:5060
[Feb 6 09:48:00] Supported: replaces, path, timer
[Feb 6 09:48:00] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 6 09:48:00] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 6 09:48:00] Content-Length: 0
[Feb 6 09:48:00]
[Feb 6 09:48:00]
[Feb 6 09:48:00] <— Transmitting SIP request (751 bytes) to UDP:10.60.10.164:5060 —>
[Feb 6 09:48:00] UPDATE sip:74500@10.60.10.164:5060 SIP/2.0
[Feb 6 09:48:00] Via: SIP/2.0/UDP 10.60.106.1:5060;rport;branch=z9hG4bKPj6a9d0d08-635a-42a9-8c79-224c9dc0c856
[Feb 6 09:48:00] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:00] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:00] Contact: sip:asterisk@10.60.106.1:5060
[Feb 6 09:48:00] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:00] CSeq: 8977 UPDATE
[Feb 6 09:48:00] Supported: 100rel, timer, replaces, norefersub
[Feb 6 09:48:00] Session-Expires: 1800;refresher=uac
[Feb 6 09:48:00] Min-SE: 90
[Feb 6 09:48:00] Max-Forwards: 70
[Feb 6 09:48:00] User-Agent: dxc
[Feb 6 09:48:00] Content-Type: application/sdp
[Feb 6 09:48:00] Content-Length: 176
[Feb 6 09:48:00]
[Feb 6 09:48:00] v=0
[Feb 6 09:48:00] o=dxc 1720138998 1720138999 IN IP4 10.60.106.1
[Feb 6 09:48:00] s=dxc
[Feb 6 09:48:00] c=IN IP4 10.60.106.1
[Feb 6 09:48:00] t=0 0
[Feb 6 09:48:00] m=audio 35016 RTP/AVP 8
[Feb 6 09:48:00] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:00] a=ptime:20
[Feb 6 09:48:00] a=maxptime:140
[Feb 6 09:48:00] a=sendrecv
[Feb 6 09:48:00]
[Feb 6 09:48:01] <— Received SIP response (795 bytes) from UDP:10.60.10.164:5060 —>
[Feb 6 09:48:01] SIP/2.0 200 OK
[Feb 6 09:48:01] Via: SIP/2.0/UDP 10.60.106.1:5060;rport=5060;branch=z9hG4bKPj6a9d0d08-635a-42a9-8c79-224c9dc0c856
[Feb 6 09:48:01] From: “Digital Station 74005” sip:74005@10.60.106.1;tag=5d0790a1-08db-4aaa-826d-529808ede8e3
[Feb 6 09:48:01] To: sip:74500@10.60.10.164;tag=1787351055
[Feb 6 09:48:01] Call-ID: 6c3d5dd5-8ba1-461c-a1b7-a0a24251fdad
[Feb 6 09:48:01] CSeq: 8977 UPDATE
[Feb 6 09:48:01] Contact: sip:74500@10.60.10.164:5060
[Feb 6 09:48:01] Supported: replaces, path, timer
[Feb 6 09:48:01] User-Agent: Grandstream GXP1782 1.0.1.126
[Feb 6 09:48:01] Session-Expires: 1800;refresher=uac
[Feb 6 09:48:01] Require: timer
[Feb 6 09:48:01] Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
[Feb 6 09:48:01] Content-Type: application/sdp
[Feb 6 09:48:01] Content-Length: 156
[Feb 6 09:48:01]
[Feb 6 09:48:01] v=0
[Feb 6 09:48:01] o=74500 8000 8001 IN IP4 10.60.10.164
[Feb 6 09:48:01] s=SIP Call
[Feb 6 09:48:01] c=IN IP4 10.60.10.164
[Feb 6 09:48:01] t=0 0
[Feb 6 09:48:01] m=audio 5004 RTP/AVP 8
[Feb 6 09:48:01] a=sendrecv
[Feb 6 09:48:01] a=rtpmap:8 PCMA/8000
[Feb 6 09:48:01] a=ptime:20
[Feb 6 09:48:01]
localhost*CLI>

20

Nothing stands out in the SIP trace, so you’d probably need to get a debug log which may show why: