Chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE

Hi all,
I see this err when call in.

This is content of file sip.confg

[general]
externip = 171.244.50.xxx
localnet = 171.244.50.xxx/255.255.255.0

Context = mobitechs
Port = 5060
Srvlookup = yes
Bindaddr = 0.0.0.0
disallow=all
Diallow = all
allow = g729
allow = g723
allow = h261
allow = h263
allow = h263p
Allow = alaw
Allow = ulaw
Allow = ilbc
Nat = 1
qualify = yes
externrefresh = 1
notifyringing = yes
notifyhold = yes
limitonpeers = yes
videosupport = no
callerid = Unknown
tos = 0x68
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints

allowguest=no

[trunk_GMSC22]
type=peer
host=10.226.2.2
context=from_trunk_GMSC
qualify=yes
;nat=no
;keepalive=45
dtmfmode=rfc2833
;disallow=all
;allow=gsm
;allow=alaw
;allow=ulaw
Canreinvite = no
insecure=port,invite

session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac

[trunk_GMSC210]
type=peer
host=10.226.2.10
context=from_trunk_GMSC
qualify=yes
;nat=no
;keepalive=45
dtmfmode=rfc2833
;disallow=all
;allow=gsm
;allow=alaw
;allow=ulaw
Canreinvite = no
insecure=port,invite

session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac


This is content of result of command : sip show peers

localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
2000/2000 (Unspecified) D N 0 UNKNOWN
2001/2001 (Unspecified) D N 0 UNKNOWN
2002/2002 (Unspecified) D N 0 UNKNOWN
trunk_GMSC210 10.226.2.10 5060 UNREACHABLE
trunk_GMSC22 10.226.2.2 N 5060 UNREACHABLE
5 sip peers [Monitored: 0 online, 5 offline Unmonitored: 0 online, 0 offline]

When the call in i see log debug

v=0
o=HuaweiSoftx3000 1076671184 1076671186 IN IP4 10.226.2.2
s=SipCall
c=IN IP4 10.226.1.132
t=0 0
m=audio 24920 RTP/AVP 18 116
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:116 telephone-event/8000
a=ptime:20
<------------->
— (12 headers 10 lines) —
Sending to 10.226.2.2:5066 (no NAT)
Using INVITE request as basis request - 0x0v2yxvybqxy4xxjbwwaujvxv2quyub@10.18.5.64
[Aug 25 12:06:29] NOTICE[23080]: chan_sip.c:22332 handle_request_invite: Unable to create/find SIP channel for this INVITE

Please re-edit your postings to mark up the logs, and configurations a preformatted text, so the forum doesn’t mangle them, and so one doesn’t have to scroll huge distances between posts.

externip ∈ localnet can never be valid!

It would help for you to explain how the failing INVITE is supposed to be identified, so we can narrow down the search for errors. Also the log fragments appear to be out of order.

There are also lots of deprecated options, which suggests you copied and pasted a configuration, without understanding it.

There is a concerning level of retransmission.

Is this a new installation, If so, why are you using a legacy channel driver, rather than chan_pjsip?

Hi Mr David551,

Thank for quick reply !
I have file sip.conf above.
The call in from GMSC1 (trunk_GMSC22) and GMSC2 (trunk_GMSC210) to my asterisk are OK.
Now, i want to make a callout from my asterisk 1.8 t0 GMSC1 & GMSC2. Help me step to step.

Note: when i use the command: sip show peers. The result is:
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
trunk_GMSC210 10.226.2.10 5060 UNREACHABLE
trunk_GMSC22 10.226.2.2 N 5060 UNREACHABLE

Thanks Mr for support

Asterisk 1.8 is almost 6 years beyond end of life! Please explain why you are using an unsupported version and a legacy SIP channel driver.

I see no evidence that you now have a sensible combination of externaddr and localnet.

Please explain your network topology? You have options that assume you are behind NAT and options that assume the opposite, so please give a detailed description of the location of NAT routers in your configuration.

Please explain how network 10/8 fists in? You have peers with private addresses, but on networks that are not explicitly declared a local.

Diallow is not a valid keyword. I’m hoping case insensitive matches are done on the other, but I don’t normally see mixed case keywords.

canreinvite was renamed as directmedia, a long time ago. I think that was even true in 1.8.

insecure=invite is meaningless here, as you have no secret. Why do you think you need insecure=port? (From what I remember before, you need to specify an explicit port number, but that doesn’t mean you need to ignore the port in matching. However not checking the port number does suggest the odd port number is not the problem.)

Your log starts halfway through the failing INVITE, so I can’t see why it isn’t being recognized. In particular, I need to see the address from which it came.
I’m not sure why OPTIONS requests are not being returned. Even if you are sending the wrong address, Asterisk always sets rport, so the peers should be able to respond to OPTIONS, even though they may be broken for actual calls.

A text book or training course is the place for step by step guides, not a peer support forum.

Thanks Mr David because have a lot Knowledge from you.
I using Asterisk 1.8 because i see it work good in some simple project.
Almost my project only process callin(from GMSC → Asterisk). But this project i need to callout(Asteris → GMSC).
About NAT: Asterisk and GMSC in the same Network so i think nat=no.
My project (call in) works good with decalre of keywork in my sip.conf.

I need callout to GMSC. I never do this until now

Hi Mr,
when i make cammand, i recived the result.

I see : From: “Anonymous” sip:Anonymous@anonymous.invalid;tag=as21a80c44
What should I do? if i want the result:
From: “199” sip:199@10.226.39.49;tag=as21a80c44

localhostCLI>
localhost
CLI>
localhost*CLI> originate SIP/76689999@10.226.2.10:5060 extension from_trunk_GMSC@199
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.226.2.10:5060:
INVITE sip:76689999@10.226.2.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Max-Forwards: 70
From: “Anonymous” sip:Anonymous@anonymous.invalid;tag=as21a80c44
To: sip:76689999@10.226.2.10:5060
Contact: sip:Anonymous@10.226.39.49:5060
Call-ID: 7058c3a1705c8ba06a3298284d97d544@10.226.39.49:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.5.0
Date: Thu, 26 Aug 2021 07:34:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1365028515 1365028515 IN IP4 10.226.39.49
s=Asterisk PBX 1.8.5.0
c=IN IP4 10.226.39.49
t=0 0
m=audio 48800 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read from UDP:10.226.2.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Call-ID: 7058c3a1705c8ba06a3298284d97d544@10.226.39.49:5060
From: "Anonymous"sip:Anonymous@anonymous.invalid;tag=as21a80c44
To: sip:76689999@10.226.2.10:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:10.226.2.10:5060 —>
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Call-ID: 7058c3a1705c8ba06a3298284d97d544@10.226.39.49:5060
From: "Anonymous"sip:Anonymous@anonymous.invalid;tag=as21a80c44
To: sip:76689999@10.226.2.10:5060;tag=z66141d4
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Got SIP response 500 “Server Internal Error” back from 10.226.2.10:5060
Transmitting (no NAT) to 10.226.2.10:5060:
ACK sip:76689999@10.226.2.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2cd850e0
Max-Forwards: 70
From: “Anonymous” sip:Anonymous@anonymous.invalid;tag=as21a80c44
To: sip:76689999@10.226.2.10:5060;tag=z66141d4
Contact: sip:Anonymous@10.226.39.49:5060
Call-ID: 7058c3a1705c8ba06a3298284d97d544@10.226.39.49:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.5.0
Content-Length: 0


<— SIP read from UDP:10.226.2.2:5066 —>
OPTIONS sip:10.226.39.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.2.2:5066;branch=z9hG4bKtrsnro1ra3okt1ong3gqnu1ss;X-DispMsg=1401
Call-ID: g1adtt1ir8q931kouti3drtsarrdq88q@10.18.5.64
From: sip:10.226.2.2:5060;tag=tn1da3ok-CC-1003-OFC-11
To: sip:10.226.39.49:5060
CSeq: 1 OPTIONS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

Hi mr david551,

I declare more in sip.conf

[isp]
type=friend
host=10.226.2.2
fromuser=199
qualify=no
context=users
insecure=very
fromdomain=10.226.39.49
realm=10.226.39.49

and i make call with: channel originate SIP/76689999@isp extension from_trunk_GMSC@199

i see: From: “Anonymous” sip:199@anonymous.invalid;tag=as4a75f080
and i want the result : From: “199” sip:199@10.226.39.49;tag=as4a75f080.

this is log:

Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.226.2.2:5060:
INVITE sip:76689999@10.226.2.2 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK46338d4d
Max-Forwards: 70
From: “Anonymous” sip:199@anonymous.invalid;tag=as4a75f080
To: sip:76689999@10.226.2.2
Contact: sip:199@10.226.39.49:5060
Call-ID: 0bbd96824a0dba541ecd7d046b70473f@10.226.39.49
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.5.0
Date: Thu, 26 Aug 2021 09:44:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1490140471 1490140471 IN IP4 10.226.39.49
s=Asterisk PBX 1.8.5.0
c=IN IP4 10.226.39.49
t=0 0
m=audio 26846 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

That is not a good reason. Please start over with Asterisk 18.6.0 and using chan_pjsip, rather than chan_sip.

The nat= setting should be left at the default unless there is a specific need to set it.

I can’t work out what on your diagram represents the machine running Asterisk, nor the netmask on any of the sub-networks.

People don’t generally use the console CLI command for serious applications, but the first thing to try would be setting the caller id in console.conf. You can also originate to a local channel and have that set the caller ID. If this is really an account name, you can use fromuser and fromdomain.

This should not have resulted in a server failure error. Nothing you can do should result in that.

Thanks Mr David551,

I have reconfig:

[trunk_GMSC22]
type=peer
host=10.226.2.2
context=from_trunk_GMSC
qualify=no
nat=no
;keepalive=45
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw
Canreinvite = no
insecure=port,invite

fromdomain=10.226.39.49
fromuser=199
usereqphone = yes
session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac

and action call from199 to 67075666668,
I see log::
From: “199” sip:199@10.226.39.49;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;user=phone

so i want to add more
From: “199” sip:199@10.226.39.49;transport=udp;user=phone;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;transport=udp;user=phone
Help me to do it.

This is full log:

------------>
Scheduling destruction of SIP dialog ‘99aisk9rdsoqtt9ookraur1uondouunn@10.18.5.64’ in 32000 ms (Method: OPTIONS)
== Setting global variable ‘QUEUE_OUTBOUND’ to ‘ADVERTISE’
== Setting global variable ‘params’ to ‘1_1_67075666668_1_199_ADVERTISE’
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.226.2.2:5060:
INVITE sip:67075666668@10.226.2.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2dd3862f
Max-Forwards: 70
From: “199” sip:199@10.226.39.49;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;user=phone
Contact: sip:199@10.226.39.49:5060
Call-ID: 1edce6b02b36b10f07ca2cc7063c23ff@10.226.39.49
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.5.0
Date: Thu, 26 Aug 2021 17:14:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1535733286 1535733286 IN IP4 10.226.39.49
s=Asterisk PBX 1.8.5.0
c=IN IP4 10.226.39.49
t=0 0
m=audio 63920 RTP/AVP 3 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read from UDP:10.226.2.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2dd3862f
Call-ID: 1edce6b02b36b10f07ca2cc7063c23ff@10.226.39.49
From: "199"sip:199@10.226.39.49;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;user=phone
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:10.226.2.2:5060 —>
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2dd3862f
Call-ID: 1edce6b02b36b10f07ca2cc7063c23ff@10.226.39.49
From: "199"sip:199@10.226.39.49;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;user=phone;tag=8o2z5h5o
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Got SIP response 500 “Server Internal Error” back from 10.226.2.2:5060
Transmitting (no NAT) to 10.226.2.2:5060:
ACK sip:67075666668@10.226.2.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5060;branch=z9hG4bK2dd3862f
Max-Forwards: 70
From: “199” sip:199@10.226.39.49;tag=as7742c6ef
To: sip:67075666668@10.226.2.2;user=phone;tag=8o2z5h5o
Contact: sip:199@10.226.39.49:5060
Call-ID: 1edce6b02b36b10f07ca2cc7063c23ff@10.226.39.49
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.5.0
Content-Length: 0


   > Channel SIP/trunk_GMSC22-00000001 was never answered.

No you don’t. You probably see

From: "199" <sip:199@10.226.39.49>;tag=as7742c6ef
To: <sip:67075666668@10.226.2.2;user=phone>

Please use </> to markup logs properly for the forum. (I’m not sure if the forum messed up the quotes, or whether that was the browser or something else.)

Specifying transport in To and From doesn’t really do anything useful for any sensible SIP UAS receiving the request. I don’t believe that Asterisk supports adding custom URI parameters, and if it isn’t adding user=phone, to From, I don’t believe there is any way of forcing it to do so, although it is possible that the automatic behaviour has changed in currently supported versions.

You are basically wasting people’s time by trying to get them to support an obsolete version of Asterisk, and you seem to be trying to communicate with a device that isn’t SIP compliant in that it is overly fussy about the format of requests. Normally the only thing that peers use From for is to use the user part as either the account name, for identification purposes, or as the caller ID. They generally do not care about the domain or any URI parameters. user=phone really is of no use if using From as the account name.

Dear Mr,
thanks for your suggestion. I used version 16.
But i have a proplem for test.
I can not press any key in sortphone with acount 6001 when call to extension 123456

Pls help me to check

I config in pjsip:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[6001]
type=endpoint
context=mobitechs
disallow=all
allow=ulaw
auth=6001
aors=6001

host = dynamic
nat = no
qualify = yes
canreinvite = no
call-limit = 1

[6001]
type=auth
auth_type=userpass
password=151080
username=6001

[6001]
type=aor
max_contacts=1

in extention file

[mobitechs]

exten => 123456,1,Playback(/myword/ivr/lochao)
;exten => 123456,1,Background(/myword/ivr/lochao)
exten => 123456,2,WaitExten(10)
exten => 1,1,Playback(/myword/ivr/lochao)
exten => 2,1,Playback(/myword/ivr/lochao)

You should specify a dtmfmode that matches that in the phone.

Also please confirm that there is no NAT and that your firewall is not blocking your selected inbound RTP port range.

I have set my sortphone RFC-2833 and config
[2000]
type=auth
auth_type=userpass
password=151080
username=2000
dtmf_mode=rfc4733

register from sortphone OK, call to extention 123456 & play file voice OK.

My server have a ip static publish and a ip local, and not firewall,
On this server, sortphone working very OK with Aterisk1.8 (RFC-2833)

I’d use the echo application, to check you had media in both directions, then use RTP debugging to see if you have telephony events, and enable the DTMF log for more details of those. If that still doesn’t give a clue, enable the full log and use “pjsip set logger on” to get a detailed SIP and SDP trace.

Thanks Mr,
My test server have IP publist 171.244.50.237 and local IP 192.168.1.127 so i set
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
local_net=192.168.1.0/24
external_media_address=171.244.50.237
external_signaling_address=171.244.50.237

tos=cs7
cos=7

So it is working now with chan_pjsip.

But now i want to use both chan_sip (5060) and chan_pjsip (5061).
I see with SIP on 5060 is OK but PJSIP on 5061 is no Media when play file voice.

I see PJSIP run on 5060 is ok, any other port is not OK.
Help me to config both SIP and PJSIP.

Thanks.

this is sip.conf
[general]
externip = 171.244.50.237
localnet = 192.168.1.127/255.255.255.0

Context = mobitechs
Port = 5060
Srvlookup = yes
Bindaddr = 0.0.0.0
disallow=all
Diallow = all
allow = g729
allow = g723
allow = h261
allow = h263
allow = h263p
Allow = alaw
Allow = ulaw
Allow = ilbc
nat = 0
qualify = yes
externrefresh = 1
notifyringing = yes
notifyhold = yes
limitonpeers = yes
videosupport = no
callerid = Unknown
tos = 0x68
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints
subscribecontext = device-hints

allowguest=no

It’s much better to run pure pjsip.

However, my guess is that the remote routers are dynamically opening the UDP ports in their NAT or firewall rules, based on inspecting the SIP traffic. but only inspect SIP on the standard port number,

Thanks for your suggestion,

I think i can fix it if check debug log.

I just upgrade from 1.8 to 16 and using PJSIP,
I have declared a endpoint [10-226-2-2] for outbound

[global]
type=global
keep_alive_interval=20

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5061
local_net=10.226.39.0/24
external_media_address=10.226.39.49
external_signaling_address=10.226.39.49

tos=cs7
cos=7

[10-226-2-2]
type=endpoint
transport=transport-udp
context=default
disallow=all
allow=ulaw
allow=alaw
allow=g726
allow=h263
aors=10-226-2-2
direct_media=yes
rewrite_contact=yes

from_domain=10.226.39.49
from_user=199

[10-226-2-2]
type=aor
contact=sip:10.226.2.2:5060

[10-226-2-2]
type=identify
endpoint=10-226-2-2
match=10.226.2.2

after that make callout:
localhost*CLI> originate PJSIP/76689999@10-226-2-2 extension from_trunk_GMSC@199
and see

From: "Anonymous"<sip:199@10.226.39.49>;tag=abadcfd2-c91a-4473-98aa-6ab48179ea48
To: <sip:76689999@10.226.2.2>

so i want to add uri look like

**From: "199"<sip:199@10.226.39.49;transport=udp;user=phone>;tag=vjyu1bqj-CC-1013-OFC-2**
**To: "76689999"<sip:76689999@10.226.2.2;transport=udp;user=phone>**

detail log

<--- Received SIP response (320 bytes) from UDP:10.226.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.226.39.49:5061;branch=z9hG4bKPjf44409ba-c32f-420c-bae3-bfc3d4007812;rport=5061
Call-ID: fe182019-d96c-4b63-94b1-b5b97dda12bc
From: "Anonymous"<sip:199@10.226.39.49>;tag=abadcfd2-c91a-4473-98aa-6ab48179ea48
To: <sip:76689999@10.226.2.2>
CSeq: 1208 INVITE
Content-Length: 0


<--- Received SIP response (348 bytes) from UDP:10.226.2.2:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 10.226.39.49:5061;branch=z9hG4bKPjf44409ba-c32f-420c-bae3-bfc3d4007812;rport=5061
Call-ID: fe182019-d96c-4b63-94b1-b5b97dda12bc
From: "Anonymous"<sip:199@10.226.39.49>;tag=abadcfd2-c91a-4473-98aa-6ab48179ea48
To: <sip:76689999@10.226.2.2>;tag=kkg1ou19
CSeq: 1208 INVITE
Content-Length: 0


<--- Transmitting SIP request (401 bytes) to UDP:10.226.2.2:5060 --->
ACK sip:76689999@10.226.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.226.39.49:5061;rport;branch=z9hG4bKPjf44409ba-c32f-420c-bae3-bfc3d4007812
From: "Anonymous" <sip:199@10.226.39.49>;tag=abadcfd2-c91a-4473-98aa-6ab48179ea48
To: <sip:76689999@10.226.2.2>;tag=kkg1ou19
Call-ID: fe182019-d96c-4b63-94b1-b5b97dda12bc
CSeq: 1208 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.20.0
Content-Length:  0

i see INVITE command from Provider to my Asterisk have transport=udp;user=phone as above in the URI. Can you tell me to know when dose this happen?