Unable to make outgoing calls to SIP provider

nimishmistry · April 21, 2021, 11:45am

I’m unable to make outgoing calls to my SIP provider.

It seem that the call goes through, but there is no ringing and the call is not placed.

Here is the call log.

Would appreciate if an expert can interpret the log and provide a solution.

<--- Received SIP request (923 bytes) from UDP:192.168.178.31:26399 --->
INVITE sip:+911234567890@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.178.31:26399;branch=z9hG4bK1200224449;rport
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
CSeq: 30 INVITE
Contact: <sip:1001@192.168.178.31:26399>
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Privacy: none
P-Preferred-Identity: <sip:1001@asterisk>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   276

v=0
o=1001 8000 8000 IN IP4 192.168.178.31
s=SIP Call
c=IN IP4 192.168.178.31
t=0 0
m=audio 48328 RTP/AVP 0 8 101
a=sendrecv
a=rtcp:48329 IN IP4 192.168.178.31
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP response (481 bytes) to UDP:192.168.178.31:26399 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.31:26399;rport=26399;received=192.168.178.31;branch=z9hG4bK1200224449
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=z9hG4bK1200224449
CSeq: 30 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1619004149/daf211c36e8dbc89216864d1edb30e77",opaque="5a05e82b2d2131a0",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (286 bytes) from UDP:192.168.178.31:26399 --->
ACK sip:+911234567890@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.178.31:26399;branch=z9hG4bK1200224449;rport
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=z9hG4bK1200224449
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
CSeq: 30 ACK
Content-Length: 0


<--- Received SIP request (1195 bytes) from UDP:192.168.178.31:26399 --->
INVITE sip:+911234567890@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.178.31:26399;branch=z9hG4bK724588732;rport
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
CSeq: 31 INVITE
Contact: <sip:1001@192.168.178.31:26399>
Authorization: Digest username="1001", realm="asterisk", nonce="1619004149/daf211c36e8dbc89216864d1edb30e77", uri="sip:+911234567890@asterisk", response="c117e09c6847c5a842aa6446c8caacdb", algorithm=md5, cnonce="12717612", opaque="5a05e82b2d2131a0", qop=auth, nc=00000006
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Privacy: none
P-Preferred-Identity: <sip:1001@asterisk>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   276

v=0
o=1001 8000 8000 IN IP4 192.168.178.31
s=SIP Call
c=IN IP4 192.168.178.31
t=0 0
m=audio 48328 RTP/AVP 0 8 101
a=sendrecv
a=rtcp:48329 IN IP4 192.168.178.31
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<--- Transmitting SIP response (306 bytes) to UDP:192.168.178.31:26399 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.31:26399;rport=26399;received=192.168.178.31;branch=z9hG4bK724588732
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>
CSeq: 31 INVITE
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (939 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj16a25cda-2dc9-4dd5-b722-d7526903d4a0
From: "Username" <sip:+919876543210@sip.provider.com>;tag=93e21ac3-0a93-4c65-98f9-a0509ad82758
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 7f82f810-e631-400b-8d8b-3c8c6783b549
CSeq: 23549 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   227

v=0
o=- 791762607 791762607 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 13616 RTP/AVP 0 101
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 request (326 bytes) from UDP:192.168.178.31:26399 --->
CANCEL sip:+911234567890@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.178.31:26399;branch=z9hG4bK724588732;rport
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
CSeq: 31 CANCEL
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.34
Content-Length: 0


<--- Transmitting SIP response (343 bytes) to UDP:192.168.178.31:26399 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.31:26399;rport=26399;received=192.168.178.31;branch=z9hG4bK724588732
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=4e467416-eabd-4b83-b00d-5cf1a77ab81a
CSeq: 31 CANCEL
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Transmitting SIP response (359 bytes) to UDP:192.168.178.31:26399 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.178.31:26399;rport=26399;received=192.168.178.31;branch=z9hG4bK724588732
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=4e467416-eabd-4b83-b00d-5cf1a77ab81a
CSeq: 31 INVITE
Server: Asterisk PBX 18.3.0
Content-Length:  0


<--- Received SIP request (304 bytes) from UDP:192.168.178.31:26399 --->
ACK sip:+911234567890@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.178.31:26399;branch=z9hG4bK724588732;rport
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=4e467416-eabd-4b83-b00d-5cf1a77ab81a
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
CSeq: 31 ACK
Content-Length: 0


<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.213:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj281c12a6-c547-4944-b2f3-9ea651711f15
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23989 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (940 bytes) to UDP:218.248.233.238:80 --->
INVITE sip:+911234567890@sip.provider.com SIP/2.0
Via: SIP/2.0/UDP 46.223.1.232:5060;rport;branch=z9hG4bKPj42d3be86-60ca-4115-849c-f426ecb57169
From: "Username" <sip:+919876543210@sip.provider.com>;tag=850e9088-f57d-43c4-992a-af5a1f66ce7d
To: <sip:+911234567890@sip.provider.com>
Contact: <sip:asterisk@46.223.1.232:5060>
Call-ID: 00958689-15bb-4c1c-8701-f66c25aeae5f
CSeq: 23990 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.3.0
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 1125190965 1125190965 IN IP4 10.8.2.5
s=Asterisk
c=IN IP4 10.8.2.5
t=0 0
m=audio 7876 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

jcolp · April 21, 2021, 11:47am

Asterisk is sending an INVITE to 218.248.233.213 port 80. There is no response. The normal SIP port is 5060, that may be your issue.

nimishmistry · April 21, 2021, 11:54am

But I haven’t configured this anywhere. How do I ensure that the request is sent to the correct port?

jcolp · April 21, 2021, 11:55am

You haven’t provided the configuration that is in use, so noone can say.

nimishmistry · April 21, 2021, 12:01pm

Sorry about that, here is my pjsip.conf. Do you also want to see the dial plan or any other configuration files?

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.178.0/24
local_net=127.0.0.1/32
external_media_address=xxx.xxx.xxx.xxx
external_signaling_address=xxx.xxx.xxx.xxx

[endpoint-internal](!)
type = endpoint
context = Long-Distance
allow = !all,g722,ulaw
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733

[auth-userpass](!)
type = auth
auth_type = userpass

[aor-single-reg](!)
type = aor
max_contacts = 1

[1001](endpoint-internal)
auth=1001
aors=1001
callerid=Username1<1001>

[1001](auth-userpass)
username=1001
password=1001

[1001](aor-single-reg)

[1002](endpoint-internal)
auth=1002
aors=1002
callerid=Username2<1002>

[1002](auth-userpass)
password=1002
username=1002

[1002](aor-single-reg)

[myprovider]
type=registration
transport=transport-udp-nat
outbound_auth=myprovider
server_uri=sip:sip.provider.com
client_uri=sip:+919876543210@sip.provider.com
contact_user=+919876543210

[myprovider]
type=auth
auth_type=userpass
username=+919876543210
password=password
realm=sip.provider.com

[myprovider]
type=aor
contact=sip:sip.provider.com

[myprovider]
type=endpoint
transport=transport-udp-nat
context=myprovider-incoming
disallow=all
allow=ulaw
outbound_auth=myprovider
aors=myprovider
direct_media=no
from_domain=sip.provider.com

[myprovider]
type=identify
endpoint=myprovider
match=xxx.xxx.xxx.xxx

jcolp · April 21, 2021, 12:03pm

If you have not specified a port in the contact URI as configured on the AOR, then it may be returned by the provider as part of SRV lookups. If so then you’d need to likely contact the provider to inquire as to why you are not getting a response, or check your networking to confirm it’s not firewalled and traffic is passing as expected.

nimishmistry · April 21, 2021, 12:34pm

Thanks for the prompt responses. Let me investigate this further.

nimishmistry · April 21, 2021, 12:37pm

Also there is a “SIP/2.0 401 Unauthorized” in the log, could this also be a problem or is this another problem?

<--- Transmitting SIP response (481 bytes) to UDP:192.168.178.31:26399 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.31:26399;rport=26399;received=192.168.178.31;branch=z9hG4bK1200224449
Call-ID: 1481145212-26399-4@BJC.BGI.BHI.DB
From: <sip:1001@asterisk>;tag=406140590
To: <sip:+911234567890@asterisk>;tag=z9hG4bK1200224449
CSeq: 30 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1619004149/daf211c36e8dbc89216864d1edb30e77",opaque="5a05e82b2d2131a0",algorithm=md5,qop="auth"
Server: Asterisk PBX 18.3.0
Content-Length:  0

jcolp · April 21, 2021, 12:37pm

That is Asterisk challenging your device to authenticate itself.

system · May 21, 2021, 12:38pm

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

Topic		Replies	Views
[solved] Asterisk as SIP UA / issues with outgoing calls Asterisk Support	1	603	June 2, 2009
Problem with outgoing call Asterisk Support	3	307	March 26, 2010
Sipcall.ch with Asterisk Outgoing calls problem Asterisk Support	1	283	July 21, 2014
Not able dial outgoing call Asterisk SIP	8	59	October 2, 2024
SIP incoming calls not working Asterisk Support	13	1414	August 3, 2012

Unable to make outgoing calls to SIP provider

Related topics