No audio is heard on Playback

VinayakVasuMV · October 9, 2024, 7:40am

No audio is heard on playback after answering the extension
public ip is 122.184.79.219 and local is 192.168.34.55


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


[1003]
type=friend
username=1003
host=dynamic
secret=XXXXX
nat=force_rport,comedia
insecure=invite,port
allow=alaw
allow=ulaw
;allow=gsm
;canreinvite=yes
nat=yes
qualify=yes
context=outbound
directmedia=no
;context=inbound-test


tecdata*CLI> originate sip/1003 extension 9895909009@outbound_airtel
  == Using SIP RTP CoS mark 5
Audio is at 12348
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 (NAT) to 152.58.202.130:38448:
INVITE sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Wed, 09 Oct 2024 07:28:55 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 717094519 717094519 IN IP4 122.184.79.219
s=Asterisk PBX 18.23.1
c=IN IP4 122.184.79.219
t=0 0
m=audio 12348 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 1003
Retransmitting #1 (NAT) to 152.58.202.130:38448:
INVITE sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.23.1
Date: Wed, 09 Oct 2024 07:28:55 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 717094519 717094519 IN IP4 122.184.79.219
s=Asterisk PBX 18.23.1
c=IN IP4 122.184.79.219
t=0 0
m=audio 12348 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:152.58.202.130:38448 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport=5060;received=122.184.79.219
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:152.58.202.130:38448 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport=5060;received=122.184.79.219
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:152.58.202.130:38448 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport=5060;received=122.184.79.219
Contact: <sip:1003@152.58.202.130:38448;transport=UDP>
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>;tag=c670c91d
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
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: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1003@152.58.202.130:38448;transport=UDP>
    -- SIP/1003-00000002 is ringing

<--- SIP read from UDP:103.183.82.56:6784 --->
REGISTER sip:122.184.79.219:33987;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;rport
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:1000@103.183.82.56:6793;transport=udp;rinstance=2ac5b46650c56f9f>
Expires: 60
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

<------------->
--- (14 headers 0 lines) ---
Sending to 103.183.82.56:6784 (NAT)

<--- Transmitting (NAT) to 103.183.82.56:6784 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;received=103.183.82.56;rport=6784
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>;tag=as47bcf942
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
CSeq: 1 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="55f2d332"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Rw_hTaxNlhLKTJOPppbwJA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:183.82.242.63:52036 --->
REGISTER sip:122.184.79.219:33972 SIP/2.0
Via: SIP/2.0/UDP 183.82.242.63:52036;rport;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421
Max-Forwards: 70
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
User-Agent: MicroSIP/3.21.4
Contact: "5000" <sip:5000@183.82.242.63:52036;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 183.82.242.63:52036 (no NAT)

<--- Transmitting (NAT) to 183.82.242.63:52036 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 183.82.242.63:52036;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421;received=183.82.242.63;rport=52036
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>;tag=as6a1e7c3f
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="5cd2d25c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1b33839df72445f7a0e668d1f3506d79' in 32000 ms (Method: REGISTER)
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=z9hG4bK658079c0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.149.27.254>;tag=as56757bc2
To: <sip:ka.ims.airtel.in>
Contact: <sip:asterisk@10.149.27.254:5060>
Call-ID: 095ca4af403279462188af1246cb8658@10.149.27.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Wed, 09 Oct 2024 07:28:57 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: 095ca4af403279462188af1246cb8658@10.149.27.254:5060
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK658079c0;rport=5060
To: <sip:ka.ims.airtel.in>;tag=65fb2b63-670630b9184a4d3f
From: "asterisk" <sip:asterisk@10.149.27.254>;tag=as56757bc2
CSeq: 102 OPTIONS
Date: Wed, 09 Oct 2024 07:28:57 GMT
Warning: 399 kka57.org "IP association no match, user not registered"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '095ca4af403279462188af1246cb8658@10.149.27.254:5060' Method: OPTIONS

<--- SIP read from UDP:152.58.202.130:38448 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK7669ee5b;rport=5060;received=122.184.79.219
Contact: <sip:1003@152.58.202.130:38448;transport=UDP>
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>;tag=c670c91d
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 356

v=0
o=Zoiper 0 1158901900 IN IP4 152.58.202.130
s=Zoiper
c=IN IP4 152.58.202.130
t=0 0
m=audio 49916 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 1158901900 and unique parts [Zoiper 0 IN IP4 152.58.202.130]
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|)
       > 0x7f46ac005f00 -- Strict RTP learning after remote address set to: 152.58.202.130:49916
Peer audio RTP is at port 152.58.202.130:49916
sip_route_dump: route/path hop: <sip:1003@152.58.202.130:38448;transport=UDP>
Transmitting (NAT) to 152.58.202.130:38448:
ACK sip:1003@152.58.202.130:38448;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK3a9b35fb;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cfc4155
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>;tag=c670c91d
Contact: <sip:anonymous@122.184.79.219:5060>
Call-ID: 43900ffd276528f9105d608e460c0a3b@122.184.79.219:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.23.1
Content-Length: 0


---
    -- SIP/1003-00000002 answered
    -- Executing [9895909009@outbound_airtel:1] SIPAddHeader("SIP/1003-00000002", "P-Preferred-Identity: <sip:+918045756630@ka.ims.airtel.in>") in new stack
    -- Executing [9895909009@outbound_airtel:2] Playback("SIP/1003-00000002", "'/var/spool/asterisk/audios/test'") in new stack
    -- <SIP/1003-00000002> Playing '/var/spool/asterisk/audios/test.slin' (language 'en')
[Oct  9 12:58:58] NOTICE[153464]: chan_sip.c:15889 sip_reregister:    -- Re-registration for  +918045756630@10.5.70.3
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.5.70.3:5060:
REGISTER sip:ka.ims.airtel.in SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK6fef97dc
Max-Forwards: 70
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 126 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 18.23.1
Authorization: Digest username="+918045756630@ka.ims.airtel.in", realm="ka.ims.airtel.in", algorithm=MD5, uri="sip:ka.ims.airtel.in", nonce="3483d140670630511YP0512c768ada563b032af5fd1774e8f9f038", response="a67d17c3b6108961cabbdd91882b3b2a", qop=auth, cnonce="6303cca0", nc=00000002
Expires: 120
Contact: <sip:s@10.149.27.254:5060>
Content-Length: 0


---

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 423 Interval Too Brief - 126
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK6fef97dc
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>;tag=1461961673
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 126 REGISTER
Min-Expires: 900
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[Oct  9 12:58:58] WARNING[153464]: chan_sip.c:24869 handle_response_register: Got 423 Interval too brief for service +918045756630@10.5.70.3, minimum is 900 seconds
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.5.70.3:5060:
REGISTER sip:ka.ims.airtel.in SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK5a1a3f88
Max-Forwards: 70
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 127 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 18.23.1
Authorization: Digest username="+918045756630@ka.ims.airtel.in", realm="ka.ims.airtel.in", algorithm=MD5, uri="sip:ka.ims.airtel.in", nonce="3483d140670630511YP0512c768ada563b032af5fd1774e8f9f038", response="493b23e31d9bf26da7c05db4c27cc1e7", qop=auth, cnonce="352becb5", nc=00000003
Expires: 900
Contact: <sip:s@10.149.27.254:5060>
Content-Length: 0


---
Really destroying SIP dialog '011bdd3813884b334a3180b16df6362e@127.0.1.1' Method: REGISTER

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK5a1a3f88
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>;tag=1999815991
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 127 REGISTER
WWW-Authenticate: Digest realm="ka.ims.airtel.in", nonce="35ae4dda670630ba1YP0bad4af0ee6fec798f6c0fdeebf6b7cb73d", stale=true, algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name 10.5.70.3
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.5.70.3:5060:
REGISTER sip:ka.ims.airtel.in SIP/2.0
Via: SIP/2.0/UDP 10.149.27.254:5060;branch=z9hG4bK3b80bddb
Max-Forwards: 70
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 128 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 18.23.1
Authorization: Digest username="+918045756630@ka.ims.airtel.in", realm="ka.ims.airtel.in", algorithm=MD5, uri="sip:ka.ims.airtel.in", nonce="35ae4dda670630ba1YP0bad4af0ee6fec798f6c0fdeebf6b7cb73d", response="38c3d431112605581d9de10218a96787", qop=auth, cnonce="15b47253", nc=00000001
Expires: 900
Contact: <sip:s@10.149.27.254:5060>
Content-Length: 0


---

<--- SIP read from UDP:10.5.70.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.149.27.254:5060;received=10.149.27.254;branch=z9hG4bK3b80bddb
From: <sip:+918045756630@ka.ims.airtel.in>;tag=as75feb83c
To: <sip:+918045756630@ka.ims.airtel.in>;tag=1926443880
Call-ID: 011bdd3813884b334a3180b16df6362e@127.0.1.1
CSeq: 128 REGISTER
Contact: <sip:s@10.149.27.254:5060;lr;transport=udp>;expires=900
Path: <sip:10.5.70.3:5060;lr;ottag=ue_term;bidx=25324168;access-type=3GPP-E-UTRAN-FDD>
Service-Route: <sip:10.5.70.3:5060;lr;bidx=0>
Service-Route: <sip:originating@skkwhfxcf30-mw.ims.airtel.in:5090;reg-id=YPJ1728408743964951;lskpmc=SYPJ;lr;ms_rest_id=SYP67056d44;pbx>
Authentication-Info: nextnonce="7565fbb0670630ba1YP0ba79f22fc523cc95e67520dae5e3bcda11"
P-Associated-URI: <sip:+918045756630@ka.ims.airtel.in>
P-Associated-URI: <tel:+918045756630>
P-Associated-URI: <sip:+9180457566!%5B3-9%5D.*!@ka.ims.airtel.in>
P-Associated-URI: <tel:+9180457566!%5B3-9%5D.*!>
P-Associated-URI: <sip:+9180457567!%5B0-1%5D.*!@ka.ims.airtel.in>
P-Associated-URI: <tel:+9180457567!%5B0-1%5D.*!>
Content-Length: 0
P-com.siemens.maximum-chat-size: 1300
P-com.siemens.maximum-IM-size: 1300
P-com.siemens.chat: direct

<------------->
--- (21 headers 0 lines) ---
[Oct  9 12:58:58] NOTICE[153464]: chan_sip.c:24961 handle_response_register: Outbound Registration: Expiry for 10.5.70.3 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '011bdd3813884b334a3180b16df6362e@127.0.1.1' Method: REGISTER

<--- SIP read from UDP:103.183.82.56:6784 --->
REGISTER sip:122.184.79.219:33987;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;rport
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:1000@103.183.82.56:6793;transport=udp;rinstance=2ac5b46650c56f9f>
Expires: 60
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

<------------->
--- (14 headers 0 lines) ---
Sending to 103.183.82.56:6784 (NAT)

<--- Transmitting (NAT) to 103.183.82.56:6784 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;received=103.183.82.56;rport=6784
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>;tag=as47bcf942
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
CSeq: 1 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="55f2d332"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Rw_hTaxNlhLKTJOPppbwJA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:183.82.242.63:52036 --->
REGISTER sip:122.184.79.219:33972 SIP/2.0
Via: SIP/2.0/UDP 183.82.242.63:52036;rport;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421
Max-Forwards: 70
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
User-Agent: MicroSIP/3.21.4
Contact: "5000" <sip:5000@183.82.242.63:52036;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 183.82.242.63:52036 (no NAT)

<--- Transmitting (NAT) to 183.82.242.63:52036 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 183.82.242.63:52036;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421;received=183.82.242.63;rport=52036
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>;tag=as6a1e7c3f
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="5cd2d25c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1b33839df72445f7a0e668d1f3506d79' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:152.58.202.130:38448 --->
REGISTER sip:192.168.99.31:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 152.58.202.130:38448;branch=z9hG4bK-524287-1---3acc19750e1192aa;rport
Max-Forwards: 70
Contact: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
To: <sip:1003@192.168.99.31;transport=UDP>
From: <sip:1003@192.168.99.31;transport=UDP>;tag=9c9d7164
Call-ID: rRBKSD9bjlP31U2cRWsBCA..
CSeq: 51 REGISTER
Expires: 60
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: Zoiper v2.10.20.4_1
Authorization: Digest username="1003",realm="127.0.0.1",nonce="6f612352",uri="sip:122.184.79.219;transport=UDP",response="6dabb2e6719fa622fec8f8cd801e8794",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.130:38448 (no NAT)
Sending to 152.58.202.130:38448 (no NAT)

<--- Transmitting (NAT) to 152.58.202.130:38448 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 152.58.202.130:38448;branch=z9hG4bK-524287-1---3acc19750e1192aa;received=152.58.202.130;rport=38448
From: <sip:1003@192.168.99.31;transport=UDP>;tag=9c9d7164
To: <sip:1003@192.168.99.31;transport=UDP>;tag=as0cfb11ff
Call-ID: rRBKSD9bjlP31U2cRWsBCA..
CSeq: 51 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="768c00a6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'rRBKSD9bjlP31U2cRWsBCA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:152.58.202.130:38448 --->
REGISTER sip:192.168.99.31:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 152.58.202.130:38448;branch=z9hG4bK-524287-1---e99bb831b85d86d4;rport
Max-Forwards: 70
Contact: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
To: <sip:1003@192.168.99.31;transport=UDP>
From: <sip:1003@192.168.99.31;transport=UDP>;tag=9c9d7164
Call-ID: rRBKSD9bjlP31U2cRWsBCA..
CSeq: 52 REGISTER
Expires: 60
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: Zoiper v2.10.20.4_1
Authorization: Digest username="1003",realm="127.0.0.1",nonce="768c00a6",uri="sip:122.184.79.219;transport=UDP",response="52c62509aaa076239395bbeeb518f9d8",algorithm=MD5
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 152.58.202.130:38448 (no NAT)
Reliably Transmitting (NAT) to 152.58.202.130:38448:
OPTIONS sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a SIP/2.0
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK04d17ad2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@122.184.79.219>;tag=as731d3ec4
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>
Contact: <sip:asterisk@122.184.79.219:5060>
Call-ID: 51b462b46a39c5e74739884b38583693@122.184.79.219:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.23.1
Date: Wed, 09 Oct 2024 07:29:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 152.58.202.130:38448 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 152.58.202.130:38448;branch=z9hG4bK-524287-1---e99bb831b85d86d4;received=152.58.202.130;rport=38448
From: <sip:1003@192.168.99.31;transport=UDP>;tag=9c9d7164
To: <sip:1003@192.168.99.31;transport=UDP>;tag=as0cfb11ff
Call-ID: rRBKSD9bjlP31U2cRWsBCA..
CSeq: 52 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>;expires=60
Date: Wed, 09 Oct 2024 07:29:01 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'rRBKSD9bjlP31U2cRWsBCA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:152.58.202.130:38448 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 122.184.79.219:5060;branch=z9hG4bK04d17ad2;rport=5060;received=122.184.79.219
Contact: <sip:152.58.202.130:38448>
To: <sip:1003@152.58.202.130:38448;transport=UDP;rinstance=73b322a117dd1d3a>;tag=c835d063
From: "asterisk" <sip:asterisk@122.184.79.219>;tag=as731d3ec4
Call-ID: 51b462b46a39c5e74739884b38583693@122.184.79.219:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
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: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '51b462b46a39c5e74739884b38583693@122.184.79.219:5060' Method: OPTIONS

<--- SIP read from UDP:103.183.82.56:6784 --->
REGISTER sip:122.184.79.219:33987;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;rport
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>
CSeq: 1 REGISTER
Max-Forwards: 70
Contact: <sip:1000@103.183.82.56:6793;transport=udp;rinstance=2ac5b46650c56f9f>
Expires: 60
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

<------------->
--- (14 headers 0 lines) ---
Sending to 103.183.82.56:6784 (NAT)

<--- Transmitting (NAT) to 103.183.82.56:6784 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:56692;branch=z9hG4bK-524287-1---bce19b916559fabe;received=103.183.82.56;rport=6784
From: <sip:1000@192.168.99.31:5060;transport=udp>;tag=99fcec2a
To: <sip:1000@192.168.99.31:5060;transport=udp>;tag=as47bcf942
Call-ID: Rw_hTaxNlhLKTJOPppbwJA..
CSeq: 1 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="55f2d332"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'Rw_hTaxNlhLKTJOPppbwJA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:183.82.242.63:52036 --->
REGISTER sip:122.184.79.219:33972 SIP/2.0
Via: SIP/2.0/UDP 183.82.242.63:52036;rport;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421
Max-Forwards: 70
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
User-Agent: MicroSIP/3.21.4
Contact: "5000" <sip:5000@183.82.242.63:52036;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 183.82.242.63:52036 (no NAT)

<--- Transmitting (NAT) to 183.82.242.63:52036 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 183.82.242.63:52036;branch=z9hG4bKPjadb8d842f81541919af67553b2ed6421;received=183.82.242.63;rport=52036
From: "5000" <sip:5000@192.168.99.31:5060>;tag=17ca9230904a494c8e91a72f8d24b05a
To: "5000" <sip:5000@192.168.99.31:5060>;tag=as6a1e7c3f
Call-ID: 1b33839df72445f7a0e668d1f3506d79
CSeq: 41051 REGISTER
Server: Asterisk PBX 18.23.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="127.0.0.1", nonce="5cd2d25c"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1b33839df72445f7a0e668d1f3506d79' in 32000 ms (Method: REGISTER)
tecdata*CLI>

david551 · October 9, 2024, 8:40am

You have a potentially serious security problem; you don’t require authentication in your outbound context, a context that can, presumably, make chargeable calls. You are also using a channel driver that is no longer supported and are showing signs of having copied and pasted its configuration. I’d suggest restarting with chan_pjsip before people try to seriously debug this.

VinayakVasuMV · October 9, 2024, 8:47am

This extension is configured in zoiper and i make the call using

tecdata*CLI> originate sip/1003 extension 9895909009@outbound_airtel

[outbound_airtel]
exten => _X.,1,Playback('/var/spool/asterisk/audios/test')
exten => _X.,n,Hangup()

david551 · October 9, 2024, 1:16pm

You also have too many retransmissions. You need to address your general network problems before looking into the specific problem.

VinayakVasuMV · October 10, 2024, 12:13pm

@david551 Network and firewall is handled by datacenter team.No audio on extension to extension calling
i have added these

ens160:  inet 192.168.34.55/24

[general]
externip=122.184.79.219
localnet=192.168.34.0/255.255.255.0

Is this because of firewall?.DC team said they opened all the ports.
Can u help to find out the cause?

system · November 9, 2024, 12:14pm

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

Topic		Replies	Views
No audio on file playback from server EC2 instance Asterisk SIP	8	1975	May 18, 2019
No audio with Skype SIP and Asterisk when playing sounds Asterisk SIP	5	499	March 28, 2019
Audio cant be heared Asterisk Support	0	217	May 15, 2007
No audio for sip calls Asterisk SIP	19	22668	March 12, 2018
Unable to hear sound supposedly being played Asterisk Support	7	3557	January 28, 2017

No audio is heard on Playback

Related topics