No audio on both sides of Airtel Sip acccount

VinayakVasuMV · October 8, 2024, 2:12pm

These are the configuration and while making a call no audio on both sides

ens160: inet 192.168.34.55/24 
ens192: inet 10.149.27.254/30 

 SBC - 10.5.70.3 

Media IPs - 
10.5.110.130
10.5.110.131
10.5.110.146
10.5.110.147
10.5.110.194
10.5.110.195
10.5.110.210
10.5.110.211

externip=122.184.79.219
localnet=192.168.34.0/24
localnet=10.5.70.0/20

[airtelsip]
disallow=all
allow=ulaw
allow=alaw
;allow=all
type=friend
dtmfmode=rfc2833
qualify=yes
nat=force_rport,comedia
insecure=invite,port
host=ka.ims.airtel.in
username=+918045756630@ka.ims.airtel.in
secret=XXXX
fromdomain=ims.airtel.in
defaultexpirey=120
canreinvite=no
context=inbound
maxexpiry=600
progressinband=yes

SIP Debugging enabled
Reliably Transmitting (NAT) to 10.5.70.3:5060:
OPTIONS sip:ka.ims.airtel.in SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK1a93ed83;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.149.27.254>;tag=as212aeb55
To: <sip:ka.ims.airtel.in>
Contact: <sip:asterisk@10.149.27.254:5060>
Call-ID: 52a2b0a566180ced7928ee1911de6441@10.149.27.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 08 Oct 2024 13:57:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: 52a2b0a566180ced7928ee1911de6441@10.149.27.254:5060
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK1a93ed83;rport=5060
To: <sip:ka.ims.airtel.in>;tag=65fb2bb0-67053a39f3f2c96
From: "asterisk" <sip:asterisk@10.149.27.254>;tag=as212aeb55
CSeq: 102 OPTIONS
Date: Tue, 08 Oct 2024 13:57:13 GMT
Warning: 399 kka57.org "IP association no match, user not registered"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '52a2b0a566180ced7928ee1911de6441@10.149.27.254:5060' Method: OPTIONS
tecdata*CLI> originate sip/1000 extension 09895909009@outbound_airtel
  == Using SIP RTP CoS mark 5
Audio is at 17118
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 103.183.82.56:6791:
INVITE sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186 SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 08 Oct 2024 13:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 741147617 741147617 IN IP4 122.184.79.219
s=Asterisk PBX 18.23.1
c=IN IP4 122.184.79.219
t=0 0
m=audio 17118 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---
    -- Called 1000
Retransmitting #1 (no NAT) to 103.183.82.56:6791:
INVITE sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186 SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 08 Oct 2024 13:57:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 741147617 741147617 IN IP4 122.184.79.219
s=Asterisk PBX 18.23.1
c=IN IP4 122.184.79.219
t=0 0
m=audio 17118 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---

<--- SIP read from UDP:103.183.82.56:6791 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9;received=122.184.79.219
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:103.183.82.56:6791 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9;received=122.184.79.219
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:103.183.82.56:6791 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9;received=122.184.79.219
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>;tag=6a64386a
CSeq: 102 INVITE
Contact: <sip:1000@103.183.82.56:6791;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,MESSAGE,OPTIONS,INFO,SUBSCRIBE
Supported: replaces,norefersub,extended-refer,timer,sec-agree,outbound,path,X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence,kpml,talk,as-feature-event
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1000@103.183.82.56:6791;transport=udp>
    -- SIP/1000-00000012 is ringing

<--- SIP read from UDP:103.183.82.56:6791 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK21faf9e9;received=122.184.79.219
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>;tag=6a64386a
CSeq: 102 INVITE
Contact: <sip:1000@103.183.82.56:6791;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,MESSAGE,OPTIONS,INFO,SUBSCRIBE
Supported: replaces,norefersub,extended-refer,timer,sec-agree,outbound,path,X-cisco-serviceuri
User-Agent: Z 5.5.15 v2.10.19.5
Allow-Events: presence,kpml,talk,as-feature-event
Content-Length: 339
Content-Type: application/sdp

v=0
o=Z 0 497355 IN IP4 103.183.82.56
s=Z
c=IN IP4 103.183.82.56
t=0 0
m=audio 6796 RTP/AVP 0 106 9 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
a=rtcp-mux
<------------->
--- (13 headers 14 lines) ---
Got SDP version 497355 and unique parts [Z 0 IN IP4 103.183.82.56]
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|gsm|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7fd5fc006090 -- Strict RTP learning after remote address set to: 103.183.82.56:6796
Peer audio RTP is at port 103.183.82.56:6796
sip_route_dump: route/path hop: <sip:1000@103.183.82.56:6791;transport=udp>
set_destination: Parsing <sip:1000@103.183.82.56:6791;transport=udp> for address/port to send to
set_destination: set destination to 103.183.82.56:6791
Transmitting (no NAT) to 103.183.82.56:6791:
ACK sip:1000@103.183.82.56:6791;transport=udp SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK2500fee1
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>;tag=6a64386a
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0


---
    -- SIP/1000-00000012 answered
    -- Executing [09895909009@outbound_airtel:1] SIPAddHeader("SIP/1000-00000012", "P-Preferred-Identity: <sip:+918045756630@ka.ims.airtel.in>") in new stack
    -- Executing [09895909009@outbound_airtel:2] Dial("SIP/1000-00000012", "SIP/airtelsip/09895909009,45,Ttg") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13446
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.5.70.3:5060:
INVITE sip:09895909009@ka.ims.airtel.in SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK1986aca3;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
To: <sip:09895909009@ka.ims.airtel.in>
Contact: <sip:asterisk@10.149.27.254:5060>
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Tue, 08 Oct 2024 13:57:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:+918045756630@ka.ims.airtel.in>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1931787324 1931787324 IN IP4 10.149.27.254
s=Asterisk PBX 18.23.1
c=IN IP4 10.149.27.254
t=0 0
m=audio 13446 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

---
    -- Called SIP/airtelsip/09895909009

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 100 Trying
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK1986aca3;rport=5060
To: <sip:09895909009@ka.ims.airtel.in>
From: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
CSeq: 102 INVITE
Date: Tue, 08 Oct 2024 13:57:19 GMT
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 183 Session Progress
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK1986aca3;rport=5060
To: <sip:09895909009@ka.ims.airtel.in>;tag=65fb2bb0-67053a3f30dfc5b1-gm-po-lucentPCSF-157308
From: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
CSeq: 102 INVITE
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-237796@10.5.70.3:5060;x-afi=19>
Content-Type: application/sdp
Content-Length: 270

v=0
o=LucentPCSF 237204755 237204755 IN IP4 i11.kka57.org
s=-
c=IN IP4 10.5.110.131
t=0 0
a=sendrecv
m=audio 38750 RTP/AVP 0 101
c=IN IP4 10.5.110.131
b=RR:3000
b=RS:1000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:30
<------------->
--- (10 headers 14 lines) ---
sip_route_dump: route/path hop: <sip:lucentNGFS-237796@10.5.70.3:5060;x-afi=19>
Got SDP version 237204755 and unique parts [LucentPCSF 237204755 IN IP4 i11.kka57.org]
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7fd604057470 -- Strict RTP learning after remote address set to: 10.5.110.131:38750
Peer audio RTP is at port 10.5.110.131:38750
    -- SIP/airtelsip-00000013 is making progress passing it to SIP/1000-00000012
       > 0x7fd604057470 -- Strict RTP switching to RTP target address 10.5.110.131:38750 as source
       > 0x7fd604057470 -- Strict RTP learning complete - Locking on source address 10.5.110.131:38750

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
To: <sip:09895909009@ka.ims.airtel.in>;tag=65fb2bb0-67053a3f30dfc5b1-gm-po-lucentPCSF-157308
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK1986aca3;rport=5060
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
CSeq: 102 INVITE
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
P-Asserted-Identity: <tel:3093>
Accept: application/sdp
Contact: <sip:lucentNGFS-237796@10.5.70.3:5060;x-afi=19>
Content-Length: 270
Content-Type: application/sdp

v=0
o=LucentPCSF 237204755 237204755 IN IP4 i11.kka57.org
s=-
c=IN IP4 10.5.110.131
t=0 0
a=sendrecv
m=audio 38750 RTP/AVP 0 101
c=IN IP4 10.5.110.131
b=RR:3000
b=RS:1000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:30
<------------->
--- (12 headers 14 lines) ---
Comparing SDP version 237204755 -> 237204755 and unique parts [LucentPCSF 237204755 IN IP4 i11.kka57.org] -> [LucentPCSF 237204755 IN IP4 i11.kka57.org]
sip_route_dump: route/path hop: <sip:lucentNGFS-237796@10.5.70.3:5060;x-afi=19>
Transmitting (NAT) to 10.5.70.3:5060:
ACK sip:lucentNGFS-237796@10.5.70.3:5060;x-afi=19 SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK6d606b68;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
To: <sip:09895909009@ka.ims.airtel.in>;tag=65fb2bb0-67053a3f30dfc5b1-gm-po-lucentPCSF-157308
Contact: <sip:asterisk@10.149.27.254:5060>
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0


---
    -- SIP/airtelsip-00000013 answered SIP/1000-00000012
    -- Channel SIP/airtelsip-00000013 joined 'simple_bridge' basic-bridge <73d6a9d3-ffdf-4b75-983b-277c07e69f82>
    -- Channel SIP/1000-00000012 joined 'simple_bridge' basic-bridge <73d6a9d3-ffdf-4b75-983b-277c07e69f82>
Really destroying SIP dialog '6HM2EqQs0l7aWlB90fl-3Q..' Method: REGISTER

<--- SIP read from UDP:10.5.70.3:5060 --->
BYE sip:asterisk@10.149.27.254:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.70.3:5060;branch=z9hG4bKe0520fd651f0703ceba4835fc5dc4e8165fb2bb0-0-75a40513-67053a4a335fde1a;_aluscr_
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_0001_1728395850-861801-189947252-LucentPCSF;e2ecallid=01d9d756
From: <sip:09895909009@ka.ims.airtel.in>;tag=65fb2bb0-67053a3f30dfc5b1-gm-po-lucentPCSF-157308
To: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
Max-Forwards: 68
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
CSeq: 1 BYE
Reason: Q.850;cause=16
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 10.5.70.3:5060 (NAT)
Scheduling destruction of SIP dialog '254f91a039f6ca967cd57e99023b37bb@ims.airtel.in' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 10.5.70.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.70.3:5060;branch=z9hG4bKe0520fd651f0703ceba4835fc5dc4e8165fb2bb0-0-75a40513-67053a4a335fde1a;_aluscr_;received=10.5.70.3;rport=5060
Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK_0001_1728395850-861801-189947252-LucentPCSF;e2ecallid=01d9d756
From: <sip:09895909009@ka.ims.airtel.in>;tag=65fb2bb0-67053a3f30dfc5b1-gm-po-lucentPCSF-157308
To: "asterisk" <sip:asterisk@ims.airtel.in>;tag=as7f13fab1
Call-ID: 254f91a039f6ca967cd57e99023b37bb@ims.airtel.in
CSeq: 1 BYE
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/airtelsip-00000013 left 'simple_bridge' basic-bridge <73d6a9d3-ffdf-4b75-983b-277c07e69f82>
    -- Channel SIP/1000-00000012 left 'simple_bridge' basic-bridge <73d6a9d3-ffdf-4b75-983b-277c07e69f82>
    -- Executing [09895909009@outbound_airtel:3] Hangup("SIP/1000-00000012", "") in new stack
  == Spawn extension (outbound_airtel, 09895909009, 3) exited non-zero on 'SIP/1000-00000012'
Scheduling destruction of SIP dialog '3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:1000@103.183.82.56:6791;transport=udp> for address/port to send to
set_destination: set destination to 103.183.82.56:6791
Reliably Transmitting (no NAT) to 103.183.82.56:6791:
BYE sip:1000@103.183.82.56:6791;transport=udp SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK57393feb
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>;tag=6a64386a
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 18.23.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:103.183.82.56:6791 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK57393feb;received=122.184.79.219
Call-ID: 3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060
From: "Anonymous"<sip:anonymous@anonymous.invalid>;tag=as28134afb
To: <sip:1000@103.183.82.56:6795;transport=udp;rinstance=39d3b926c6019186>;tag=6a64386a
CSeq: 103 BYE
Contact: <sip:1000@103.183.82.56:6791;transport=udp>
User-Agent: Z 5.5.15 v2.10.19.5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3a6d8cdc0461d77938e44f8302df68ae@122.184.79.219:5060' Method: INVITE
tecdata*CLI>

EkFudrek · October 8, 2024, 2:18pm

Could it be that you are using the obsolete chan_sip stack using obsolete NAT and rather dangerous settings?

Inderjeet01 · October 8, 2024, 3:07pm

Here are a few things to verify:
1.Check the firewall or iptables rules for rtp ports.
2.Ensure you’re using the same codec that the SBC offers.
3.Confirm that the media server’s IP (10.5.110.131) is allowed in the routing via the gateway of customer’s ip

VinayakVasuMV · October 8, 2024, 3:12pm

i already check all these and they are okay

Destination     Gateway         Genmask         Flags Metric Ref    Use Iface
0.0.0.0         192.168.34.1    0.0.0.0         UG    0      0        0 ens160
0.0.0.0         10.149.27.253   0.0.0.0         UG    0      0        0 ens192
10.5.70.3       10.149.27.253   255.255.255.255 UGH   0      0        0 ens192
10.5.110.0      10.149.27.253   255.255.255.0   UG    0      0        0 ens192
10.149.27.252   0.0.0.0         255.255.255.252 U     0      0        0 ens192
192.168.34.0    0.0.0.0         255.255.255.0   U     0      0        0 ens160

Inderjeet01 · October 8, 2024, 3:23pm

@VinayakVasuMV
Before routing the call to the extension, you can route it to an audio playback first to confirm if the issue is with the extension not receiving voice.

VinayakVasuMV · October 9, 2024, 4:09am

@david551 @jcolp Any help on this issue?

jcolp · October 9, 2024, 8:23am

Please don’t tag people who aren’t participating.

system · November 8, 2024, 8:24am

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

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