Asterisk PJSIP no voice on incoming calls

elesh.medari · July 26, 2023, 8:19pm

Hi, I am having issue with the asterisk pjsip. I receive the call, but I don’t get any voice. Asterisk is deployed in kubernetes, we opened the ports 5060 for outbound communication to SIP servers and 10000-20000 ports for inbound communication from SIP servers.

pjsip.conf

[global]
type = global
max_initial_qualify_time = 4

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = 135.84.XX.XXX (External IP of the Kubernetes cluster)
external_signaling_address = 135.84..XX.XXX (External IP of the Kubernetes cluster)
local_net = 10.YYY.YY.0/24 (Internal IP of Kubernetes node)
local_net = 10.42.0.0/24 (Internal IP of pod)
allow_reload = yes

[sip-outbound]
type = registration
transport = transport-udp
outbound_auth = sip-outbound
server_uri = sip:gw1.sip.us
client_uri = sip:USERNAME@gw1.sip.us  (USERNAME to connect to SIP Servers)
contact_user = USERNAME  (USERNAME to connect to SIP Servers)

[sip-outbound]
type = aor
contact = sip:USERNAME@gw1.sip.us:5060  (USERNAME to connect to SIP Servers)
qualify_frequency = 30

[sip-outbound]
type = identify
endpoint = sip-outbound
match = gw1.sip.us

[sip-outbound]
type = auth
auth_type = userpass
username = USERNAME (USERNAME to connect to SIP Servers)
password = PASSWORD (PASSWORD to connect to SIP Servers)
realm = gw1.sip.us

[sip-outbound]
type = endpoint
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = t140
timers = no
trust_id_inbound = yes
ice_support = yes
from_domain = gw1.sip.us
auth = sip-outbound
outbound_auth = sip-outbound
aors = sip-outbound
media_use_received_transport = yes
media_encryption = no
rtp_symmetric = yes
force_rport = no
rewrite_contact = yes
direct_media = no

[sip-outbound]
type = peer
insecure = port,invite
host = gw1.sip.us
port = 5060
dtmf_mode = rfc2833
canreinvite = no
disallow = all
allow = ulaw
qualify = yes
qualifyfreq = 30
nat = force_rport,comedia
trustrpid = yes
fromdomain = gw1.sip.us
username = USERNAME (USERNAME to connect to SIP Servers)
secret = PASSWORD (PASSWORD to connect to SIP Servers)
context = from-trunk
rfc2833compensate = yes
session-timers = refuse
ice_support = yes
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
rtpkeepalive = 2
direct_media = no

[sip-outbound]
type = peer
insecure = port,invite
host = gw2.sip.us
port = 5060
dtmf_mode = rfc2833
canreinvite = no
disallow = all
allow = ulaw
qualify = yes
qualifyfreq = 30
nat = force_rport,comedia
trustrpid = yes
fromdomain = gw2.sip.us
username = USERNAME (USERNAME to connect to SIP Servers)
secret = PASSWORD (PASSWORD to connect to SIP Servers)
context = from-trunk
rfc2833compensate = yes
session-timers = refuse
ice_support = yes
rtp_symmetric = yes
force_rport = no
rewrite_contact = yes
rtpkeepalive = 2
direct_media = no

I see the log says the playback

[Jul 26 20:12:14]     -- <Local/15707024636@default-00000018;1> Playing 'digits/3.gsm' (language 'ulti-en-US')
[Jul 26 20:12:15]     -- <Local/15707024636@default-00000018;1> Playing 'digits/4.gsm' (language 'ulti-en-US')
[Jul 26 20:12:15]     -- <Local/15707024636@default-00000018;1> Playing 'digits/5.gsm' (language 'ulti-en-US')

But I don’t hear any voice in my cell phone.

Can you please let me know what is the issue?

jcolp · July 26, 2023, 8:40pm

Examine the actual network traffic and see where it is going. You can see the SIP signaling using “pjsip set logger on” and the RTP traffic using “rtp set debug on”. If media is leaving Asterisk as expected, then your problem is most likely outside of Asterisk.

david551 · July 26, 2023, 9:45pm

I find the distinction between inbound and outbound confusing, here; both of these ports/ranges need to be open in both directions.

Also I’m not sure if you are using SIP servers in the correct sense, as all incoming calls come from SIP clients.

USERNAME shouldn’t relate to the SIP registrar, it should be the extension that handles incoming calls. It might look the same, but it has a different meaning.

This should not be in pjsip.conf.

Why? It isn’t even consistent with your sip.conf style settings. It is invalid in sip.conf.

Bad choice of endpoint name. type=identify is only valid when acting as server for initial requests, but outbound implies acting as client.

There is no registration in your sip.conf part, so it is not equivalent to the pjsip part.

elesh.medari · July 26, 2023, 10:38pm

Hi Sorry… I am a newbie to the asterisk side.
Yes we opened the 2 way communication to the SIP Servers and ports 10000 to 20000 ports. I updated the PJSIP.conf as you suggested. Now it looks like below

[global]
type = global
max_initial_qualify_time = 4

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = 135.84.XX.XXX (External IP of the Kubernetes cluster)
external_signaling_address = 135.84..XX.XXX (External IP of the Kubernetes cluster)
local_net = 10.YYY.YY.0/24 (Internal IP of Kubernetes node)
local_net = 10.42.0.0/24 (Internal IP of pod)
allow_reload = yes

[sip-outbound]
type = registration
transport = transport-udp
outbound_auth = sip-outbound
server_uri = sip:gw1.sip.us
client_uri = sip:USERNAME@gw1.sip.us  (USERNAME to connect to SIP Servers)

[sip-outbound]
type = aor
contact = sip:USERNAME@gw1.sip.us:5060  (USERNAME to connect to SIP Servers)
qualify_frequency = 30

[sip-outbound]
type = identify
endpoint = sip-outbound
match = gw1.sip.us

[sip-outbound]
type = auth
auth_type = userpass
username = USERNAME (USERNAME to connect to SIP Servers)
password = PASSWORD (PASSWORD to connect to SIP Servers)
realm = gw1.sip.us

[sip-outbound]
type = endpoint
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = t140
timers = no
trust_id_inbound = yes
ice_support = yes
from_domain = gw1.sip.us
auth = sip-outbound
outbound_auth = sip-outbound
aors = sip-outbound
media_use_received_transport = yes
media_encryption = no
rtp_symmetric = yes
rewrite_contact = yes
direct_media = no

Now I see

WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'

[Jul 26 22:32:17] WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'

[Jul 26 22:33:49] WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'

[Jul 26 22:35:21] WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'

[Jul 26 22:36:53] WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'

david551 · July 26, 2023, 11:07pm

I can’t see any change in your registration settings that would stop the registration getting through.l

elesh.medari · July 27, 2023, 2:54pm

The registration is happening when I run the same code locally. Not sure what happened but I am seeing this in the Kubernetes

Contact sip-outbound/sip:5252323987@gw1.sip.us:5060 is now Unreachable. RTT: 0.000 msec

Below are the PJSIP logs

<--- Transmitting SIP request (541 bytes) to UDP:65.254.44.194:5060 --->
REGISTER sip:gw1.sip.us SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjXTxu-ZqSXeOBkqcDt8fbAp4KZTYb7h50
From: <sip:5252323987@gw1.sip.us>;tag=MsldP5WgQ-EU-t5bnOvY9MoS-sOXLPjn
To: <sip:5252323987@gw1.sip.us>
Call-ID: b7lpxd6qXefgxDl9IbGaWlWyNHffKzdN
CSeq: 56161 REGISTER
Contact: <sip:5252323987@135.84.76.230:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjyBl0jx.CBJZ6ccNmVl5U9ZWbcvGHA.Gd
From: <sip:sip-outbound@gw1.sip.us>;tag=At-9SNMJrmjGpy-cRW55VPne38Qc3gTo
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: hQytgPFO0RACgCpWSoXpAiPBXRpEG1nL
CSeq: 32845 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (541 bytes) to UDP:65.254.44.194:5060 --->
REGISTER sip:gw1.sip.us SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjXTxu-ZqSXeOBkqcDt8fbAp4KZTYb7h50
From: <sip:5252323987@gw1.sip.us>;tag=MsldP5WgQ-EU-t5bnOvY9MoS-sOXLPjn
To: <sip:5252323987@gw1.sip.us>
Call-ID: b7lpxd6qXefgxDl9IbGaWlWyNHffKzdN
CSeq: 56161 REGISTER
Contact: <sip:5252323987@135.84.76.230:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjyBl0jx.CBJZ6ccNmVl5U9ZWbcvGHA.Gd
From: <sip:sip-outbound@gw1.sip.us>;tag=At-9SNMJrmjGpy-cRW55VPne38Qc3gTo
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: hQytgPFO0RACgCpWSoXpAiPBXRpEG1nL
CSeq: 32845 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (541 bytes) to UDP:65.254.44.194:5060 --->
REGISTER sip:gw1.sip.us SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjXTxu-ZqSXeOBkqcDt8fbAp4KZTYb7h50
From: <sip:5252323987@gw1.sip.us>;tag=MsldP5WgQ-EU-t5bnOvY9MoS-sOXLPjn
To: <sip:5252323987@gw1.sip.us>
Call-ID: b7lpxd6qXefgxDl9IbGaWlWyNHffKzdN
CSeq: 56161 REGISTER
Contact: <sip:5252323987@135.84.76.230:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (541 bytes) to UDP:65.254.44.194:5060 --->
REGISTER sip:gw1.sip.us SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjXTxu-ZqSXeOBkqcDt8fbAp4KZTYb7h50
From: <sip:5252323987@gw1.sip.us>;tag=MsldP5WgQ-EU-t5bnOvY9MoS-sOXLPjn
To: <sip:5252323987@gw1.sip.us>
Call-ID: b7lpxd6qXefgxDl9IbGaWlWyNHffKzdN
CSeq: 56161 REGISTER
Contact: <sip:5252323987@135.84.76.230:5060>
Expires: 3600
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


[Jul 27 14:52:20] WARNING[42]: res_pjsip_outbound_registration.c:905 schedule_retry: No response received from 'sip:gw1.sip.us' on registration attempt to 'sip:5252323987@gw1.sip.us', retrying in '60'
<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjt.DBwsYpot9KK2GtptchUwfk9TTGMSfk
From: <sip:sip-outbound@gw1.sip.us>;tag=pBii7kGIn6LB65dJdewPhyW8eLpM28rH
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: AULS5fMGBRwD.zKoe6v54eikbT1z75Np
CSeq: 65409 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjpTNQtGUJa0lvUBbU6kTF7TfZWP1FPuNb
From: <sip:sip-outbound@gw1.sip.us>;tag=tKZOd6kByLJqBAKT4KNIunBG4j9PN4wp
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: Jz26At-w5bHB7ISWCkluagxY0iwhJxGY
CSeq: 31672 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjpTNQtGUJa0lvUBbU6kTF7TfZWP1FPuNb
From: <sip:sip-outbound@gw1.sip.us>;tag=tKZOd6kByLJqBAKT4KNIunBG4j9PN4wp
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: Jz26At-w5bHB7ISWCkluagxY0iwhJxGY
CSeq: 31672 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjpTNQtGUJa0lvUBbU6kTF7TfZWP1FPuNb
From: <sip:sip-outbound@gw1.sip.us>;tag=tKZOd6kByLJqBAKT4KNIunBG4j9PN4wp
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: Jz26At-w5bHB7ISWCkluagxY0iwhJxGY
CSeq: 31672 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjRVf5AoHkvEqMIIeigGJxJuDkJl9PwpFF
From: <sip:sip-outbound@gw1.sip.us>;tag=NV2mIwhHavYp4GSg7KD7kZ85yHocFQAA
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: KYpcYumbLGni3j8bHhRdGX5H26DXhZBl
CSeq: 16882 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0

jcolp · July 27, 2023, 2:58pm

It’s not getting any SIP responses, and since the different factor is Kubernetes it’s probably something related to networking and that.

david551 · July 27, 2023, 3:00pm

All the logs tell you is that you are not getting responses from 65.254.44.194. The only thing that would cause that in Asterisk is if that is not the right address. Otherwise it is a network or remote end problem, where network includes the container system.

elesh.medari · July 27, 2023, 7:04pm

I fixed the network now. Can you please check this logs please and let me know what the issue is? I see
Sent RTP packet to, but I don’t see the packet being received from SIP.

Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13211 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, histinfo
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Type: application/sdp
Content-Length:   459

v=0
o=- 389524449 389524449 IN IP4 135.84.76.230
s=Asterisk
c=IN IP4 135.84.76.230
t=0 0
m=audio 17706 RTP/AVP 0 101
a=ice-ufrag:0095c46648b66862236b3dc34167c959
a=ice-pwd:09dfa09a4ec5ee1625d5c37717365ee9
a=candidate:Ha2a1f61 1 UDP 2130706431 10.42.31.97 17706 typ host
a=candidate:Ha2a1f61 2 UDP 2130706430 10.42.31.97 17707 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (332 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 135.84.76.230:5060;rport=49182;branch=z9hG4bKPj6qbJ.ASw4XAwtkB2gfsaojQsTz7nmpqv;received=135.84.76.230
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13211 INVITE
Content-Length: 0


<--- Received SIP response (530 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 135.84.76.230:5060;rport=49182;branch=z9hG4bKPj6qbJ.ASw4XAwtkB2gfsaojQsTz7nmpqv;received=135.84.76.230
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=cf2756ab99a8b0ac875a25a28d35ded3.84a87765
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13211 INVITE
Proxy-Authenticate: Digest realm="gw1.sip.us", nonce="ZMLBAWTCvql+itScVPkfRmlqX4UV/NyYKmvj7HevtTkIo59AMVCawmOFvV/C", qop="auth"
Content-Length: 0


<--- Transmitting SIP request (424 bytes) to UDP:65.254.44.194:5060 --->
ACK sip:15707024636@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPj6qbJ.ASw4XAwtkB2gfsaojQsTz7nmpqv
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=cf2756ab99a8b0ac875a25a28d35ded3.84a87765
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13211 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Transmitting SIP request (1373 bytes) to UDP:65.254.44.194:5060 --->
INVITE sip:15707024636@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjE4OLzU.k829oVjtqDWcstaZs0Nae4pfb
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>
Contact: <sip:asterisk@135.84.76.230:5060>
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, histinfo
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Proxy-Authorization: Digest username="5252323987", realm="gw1.sip.us", nonce="ZMLBAWTCvql+itScVPkfRmlqX4UV/NyYKmvj7HevtTkIo59AMVCawmOFvV/C", uri="sip:15707024636@gw1.sip.us:5060", response="abd06a1d99b58be4865db4c1fae3173d", cnonce="yhBOjGJaHdlstIjNBbQ3jnI1PACJu6F", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   459

v=0
o=- 389524449 389524449 IN IP4 135.84.76.230
s=Asterisk
c=IN IP4 135.84.76.230
t=0 0
m=audio 17706 RTP/AVP 0 101
a=ice-ufrag:0095c46648b66862236b3dc34167c959
a=ice-pwd:09dfa09a4ec5ee1625d5c37717365ee9
a=candidate:Ha2a1f61 1 UDP 2130706431 10.42.31.97 17706 typ host
a=candidate:Ha2a1f61 2 UDP 2130706430 10.42.31.97 17707 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (332 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 135.84.76.230:5060;rport=49182;branch=z9hG4bKPjE4OLzU.k829oVjtqDWcstaZs0Nae4pfb;received=135.84.76.230
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 INVITE
Content-Length: 0


<--- Received SIP response (1088 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 135.84.76.230:5060;received=135.84.76.230;rport=49182;branch=z9hG4bKPjE4OLzU.k829oVjtqDWcstaZs0Nae4pfb
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 INVITE
Record-Route: <sip:65.254.44.194:5060;lr=on;ftag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7;vsf=AAAAAAgNBAMHAgUBBwJ5JxBGH10aGV5bBnM-;dlgcor=4021.d371;proxy_media=yes>
Contact: <sip:15707024636@67.231.1.170:5060>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
P-Asserted-Identity: "MEDARI ELESH" <tel:+15707024636;npdi;enumdi>, "MEDARI ELESH" <sip:+15707024636@65.254.44.194;user=phone>
Content-Length:   277
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 705447 171363 IN IP4 65.254.44.194
s=SIP Media Capabilities
c=IN IP4 65.254.44.194
t=0 0
a=rtpengine:e77aee1d7d44
m=audio 61934 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:61935
a=ptime:20

<--- Received SIP response (1088 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 135.84.76.230:5060;received=135.84.76.230;rport=49182;branch=z9hG4bKPjE4OLzU.k829oVjtqDWcstaZs0Nae4pfb
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 INVITE
Record-Route: <sip:65.254.44.194:5060;lr=on;ftag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7;vsf=AAAAAAgNBAMHAgUBBwJ5JxBGH10aGV5bBnM-;dlgcor=4021.d371;proxy_media=yes>
Contact: <sip:15707024636@67.231.1.170:5060>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
P-Asserted-Identity: "MEDARI ELESH" <tel:+15707024636;npdi;enumdi>, "MEDARI ELESH" <sip:+15707024636@65.254.44.194;user=phone>
Content-Length:   277
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 705447 171363 IN IP4 65.254.44.194
s=SIP Media Capabilities
c=IN IP4 65.254.44.194
t=0 0
a=rtpengine:e77aee1d7d44
m=audio 61934 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:61935
a=ptime:20

<--- Received SIP response (992 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 135.84.76.230:5060;received=135.84.76.230;rport=49182;branch=z9hG4bKPjE4OLzU.k829oVjtqDWcstaZs0Nae4pfb
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 INVITE
Record-Route: <sip:65.254.44.194:5060;lr=on;ftag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7;vsf=AAAAAAgNBAMHAgUBBwJ5JxBGH10aGV5bBnM-;dlgcor=4021.d371;proxy_media=yes>
Accept: application/sdp
Contact: <sip:15707024636@67.231.1.170:5060>
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Supported: replaces
Content-Length:   277
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 705447 171363 IN IP4 65.254.44.194
s=SIP Media Capabilities
c=IN IP4 65.254.44.194
t=0 0
a=rtpengine:e77aee1d7d44
m=audio 61934 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:61935
a=ptime:20

<--- Transmitting SIP request (543 bytes) to UDP:65.254.44.194:5060 --->
ACK sip:15707024636@67.231.1.170:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjo0trusgAaEuvvhThhicuueh3tYNDSMkY
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13212 ACK
Route: <sip:65.254.44.194:5060;lr;ftag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7;vsf=AAAAAAgNBAMHAgUBBwJ5JxBGH10aGV5bBnM-;dlgcor=4021.d371;proxy_media=yes>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0





<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjENr53XshOjPuyySyWDEhFxySaykRRtAR
From: <sip:sip-outbound@gw1.sip.us>;tag=KNbZrjb9Ti2ld3dmJPdfF.2ieKzp.UT.
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: pX3OPna2pnDUwaneC7tGq7EqtuUeaAj0
CSeq: 56901 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012152, ts 164880, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012153, ts 165040, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012154, ts 165200, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012155, ts 165360, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012156, ts 165520, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012157, ts 165680, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012158, ts 165840, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012159, ts 166000, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012160, ts 166160, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012161, ts 166320, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012162, ts 166480, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012163, ts 166640, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012164, ts 166800, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012165, ts 166960, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012166, ts 167120, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012167, ts 167280, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012168, ts 167440, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012169, ts 167600, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012170, ts 167760, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012171, ts 167920, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012172, ts 168080, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012173, ts 168240, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012174, ts 168400, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012175, ts 168560, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012176, ts 168720, len 000160)
<--- Transmitting SIP request (433 bytes) to UDP:65.254.44.194:5060 --->
OPTIONS sip:5252323987@gw1.sip.us:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjENr53XshOjPuyySyWDEhFxySaykRRtAR
From: <sip:sip-outbound@gw1.sip.us>;tag=KNbZrjb9Ti2ld3dmJPdfF.2ieKzp.UT.
To: <sip:5252323987@gw1.sip.us>
Contact: <sip:sip-outbound@135.84.76.230:5060>
Call-ID: pX3OPna2pnDUwaneC7tGq7EqtuUeaAj0
CSeq: 56901 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012177, ts 168880, len 000160)
Sent RTP packet to      65.254.44.194:61934 (type 00, seq 012178, ts 169040, len 000160)
<--- Received SIP response (375 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 135.84.76.230:5060;rport=49182;branch=z9hG4bKPjENr53XshOjPuyySyWDEhFxySaykRRtAR;received=135.84.76.230
From: <sip:sip-outbound@gw1.sip.us>;tag=KNbZrjb9Ti2ld3dmJPdfF.2ieKzp.UT.
To: <sip:5252323987@gw1.sip.us>;tag=cf2756ab99a8b0ac875a25a28d35ded3.4eec329a
Call-ID: pX3OPna2pnDUwaneC7tGq7EqtuUeaAj0
CSeq: 56901 OPTIONS
Content-Length: 0


[Jul 27 19:00:23] NOTICE[111][C-00000002]: pbx_spool.c:463 attempt_thread: Call completed to Local/15707024636
<--- Transmitting SIP request (567 bytes) to UDP:65.254.44.194:5060 --->
BYE sip:15707024636@67.231.1.170:5060 SIP/2.0
Via: SIP/2.0/UDP 135.84.76.230:5060;rport;branch=z9hG4bKPjxh140Sx71CX4uVnuiVof6Eg4mRClpDJw
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13213 BYE
Route: <sip:65.254.44.194:5060;lr;ftag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7;vsf=AAAAAAgNBAMHAgUBBwJ5JxBGH10aGV5bBnM-;dlgcor=4021.d371;proxy_media=yes>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.1
Content-Length:  0


<--- Received SIP response (340 bytes) from UDP:65.254.44.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 135.84.76.230:5060;received=135.84.76.230;rport=49182;branch=z9hG4bKPjxh140Sx71CX4uVnuiVof6Eg4mRClpDJw
From: <sip:18004321729@gw1.sip.us>;tag=yFIbnKtW9A0tbcJ0gqLcOodujqPmy-b7
To: <sip:15707024636@gw1.sip.us>;tag=gK08c76b5c
Call-ID: sCuYPg6b3mE0qvNst99vE70SgwCS.uTm
CSeq: 13213 BYE
Content-Length: 0

elesh.medari · July 28, 2023, 4:37pm

Currently we see the alias registered to port 38052. The SIP support says the alias should also point to port 5060. Is there a configuration in pjsip.conf to force it to use port 5060?

Screenshot 2023-07-28 at 12.35.32 PM

jcolp · July 28, 2023, 5:04pm

“alias” is not something Asterisk adds.

elesh.medari · July 28, 2023, 5:12pm

Then where does it come from? From firewall?

jcolp · July 28, 2023, 5:14pm

A SIP proxy probably.

elesh.medari · July 28, 2023, 7:10pm

I reached out to SIP, they said they won’t control this and its something comes from the configuration. Is it something I need to do in PJSIP.CONF or anywhere else

They say we wont here voice until left and right side ports are same.
Screenshot 2023-07-28 at 3.09.12 PM

jcolp · July 28, 2023, 8:03pm

Well, that’s not due to Asterisk. If the source UDP port is different then that would be due to something network related - such as a NAT using a different external port.

elesh.medari · July 28, 2023, 9:30pm

Just to add some more context into the traffic flow here. We have asterisk deployed on K8s.
K8 POD internal IPs (10.42.XX.XX). → that get’s SNAT’ed to 10.150.92.x at the K8 Node Gateway → internal firewall with SIP ALG enabled and no NATing → Edge firewall where this 10.150.92.x is SNATed to 135.84.76.x when reaching the SIP server [65.254.44.194] and also this firewall has the SIP ALG enabled. We do see the outgoing traffic is NATed as expected but, when there is a new session initiated from SIP server to our source, somehow Juniper SRX is NATing to the POD internal IP for which this edge firewall doesn’t see as the source at any point in time. We do believe this could be because of some header information being passed by SIP provider. Same firewalls are in place with VM based deployment outside the K8s. But its working perfectly. The issue is only with K8s. With this setup into consideration any recommendations in changing the configuration to get the voice in the incoming calls.
Any help will be appreciated.

david551 · July 28, 2023, 11:48pm

This is strongly discouraged.

system · August 27, 2023, 11:49pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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