SIP Softphone works , Asterisk does not

Asterisk Asterisk SIP
1

I have an account on IncrediblePBX on a Public IP .

I can register a softphone or hard phone to that account and make and receive calls.

When I try to use Asterisk on Raspberry Pi from the same location (shutting down hard phone and softphone) as the softphone or hardphone I can register, and receive calls but outbound calls show
SIP/2.0 401 Unauthorized

I have tried oh so many variations in sip.conf that it makes my head spin, and could not possibly post them all.

Currently I have something like this

[general]
…
…

register => 7870:Password@my.domain.com/7870

[7870]
; insecure=invite,port
; insecure=invite
insecure=invite
dtmfmode=rfc2833
secret=Password
; type=peer
type=friend
nat=yes
qualify=yes
keepalive=45
monitor=yes
; callerid=“7870” <7870>
host=my.domain.com ; This device needs to register
directmedia=no ; Typically set to NO if behind NAT
disallow=all
fromuser=7870
authuser=7870
username=7870
;fromdomain=my.domain.com
;allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
context=from-pstn

testing with ths dial plan primarily
exten => 411,1,Dial(SIP/1800373341160@7870,1)
exten => 411,2,Hangup()

I have tested without the “,1” (timeout?) in line 1

2 
can we see the configuration of extensions.conf file?

SANOGO Touna

3

This is what I have been testing with primarily

exten => 411,1,Dial(SIP/1800373341160@7870,1)
exten => 411,2,Hangup()

I have tested without the “,1” (timeout?) in line 1

4

Show please the SIP trace for the above reply

sip set debug on

5

Reliably Transmitting (NAT) to 199.199.199.199:5060:
OPTIONS sip:my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK54de31c8;rport
Max-Forwards: 70
From: “asterisk” sip:7870@192.168.1.50;tag=as7085f995
To: sip:my.domain.com
Contact: sip:7870@192.168.1.50:5060
Call-ID: 36022dff3fd5334e754f9ed314cc431c@192.168.1.50:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Date: Mon, 22 Jul 2019 00:43:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:199.199.199.199:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK54de31c8;received=192.168.1.50;rport=5060
From: “asterisk” sip:7870@192.168.1.50:5060;tag=as7085f995
To: sip:my.domain.com;tag=as33cf13ad
Call-ID: 36022dff3fd5334e754f9ed314cc431c@192.168.1.50:5060
CSeq: 102 OPTIONS
Server: YoMaMa
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:199.199.199.199:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘36022dff3fd5334e754f9ed314cc431c@192.168.1.50:5060’ Method: OPTIONS
== Using SIP RTP CoS mark 5
> 0x3048d38 – Strict RTP learning after remote address set to: 192.168.1.49:13456
– Executing [411@from-internal:1] Dial(“SIP/5901-0000000d”, “SIP/1800373341160@7870,1”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 15560
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 199.199.199.199:5060:
INVITE sip:1800373341160@my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4dfbbc73;rport
Max-Forwards: 70
From: “UTSTAR1” sip:7870@192.168.1.50;tag=as54f61e8c
To: sip:1800373341160@my.domain.com
Contact: sip:7870@192.168.1.50:5060
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Date: Mon, 22 Jul 2019 00:43:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 302216126 302216126 IN IP4 192.168.1.50
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 192.168.1.50
t=0 0
m=audio 15560 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

-- Called SIP/1800373341160@7870

<— SIP read from UDP:199.199.199.199:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4dfbbc73;received=192.168.1.50;rport=5060
From: “UTSTAR1” sip:7870@192.168.1.50:5060;tag=as54f61e8c
To: sip:1800373341160@my.domain.com;tag=as2f73c1dd
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 102 INVITE
Server: YoMaMa
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6df06d37”
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 199.199.199.199:5060:
ACK sip:1800373341160@my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK4dfbbc73;rport
Max-Forwards: 70
From: “UTSTAR1” sip:7870@192.168.1.50;tag=as54f61e8c
To: sip:1800373341160@my.domain.com;tag=as2f73c1dd
Contact: sip:7870@192.168.1.50:5060
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Content-Length: 0

Audio is at 15560
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 199.199.199.199:5060:
INVITE sip:1800373341160@my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK59b1364e;rport
Max-Forwards: 70
From: “UTSTAR1” sip:7870@192.168.1.50;tag=as54f61e8c
To: sip:1800373341160@my.domain.com
Contact: sip:7870@192.168.1.50:5060
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Authorization: Digest username=“7870”, realm=“asterisk”, algorithm=MD5, uri="sip:1800373341160@my.domain.com", nonce=“6df06d37”, response=“50f98f21acb736da31028d6e5f7acd93”
Date: Mon, 22 Jul 2019 00:43:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 302216126 302216127 IN IP4 192.168.1.50
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 192.168.1.50
t=0 0
m=audio 15560 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

Retransmitting #1 (NAT) to 199.199.199.199:5060:
INVITE sip:1800373341160@my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK59b1364e;rport
Max-Forwards: 70
From: “UTSTAR1” sip:7870@192.168.1.50;tag=as54f61e8c
To: sip:1800373341160@my.domain.com
Contact: sip:7870@192.168.1.50:5060
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Authorization: Digest username=“7870”, realm=“asterisk”, algorithm=MD5, uri="sip:1800373341160@my.domain.com", nonce=“6df06d37”, response=“50f98f21acb736da31028d6e5f7acd93”
Date: Mon, 22 Jul 2019 00:43:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 302216126 302216127 IN IP4 192.168.1.50
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 192.168.1.50
t=0 0
m=audio 15560 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

<— SIP read from UDP:199.199.199.199:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK59b1364e;received=192.168.1.50;rport=5060
From: “UTSTAR1” sip:7870@192.168.1.50:5060;tag=as54f61e8c
To: sip:1800373341160@my.domain.com;tag=as2f73c1dd
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 103 INVITE
Server: YoMaMa
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Transmitting (NAT) to 199.199.199.199:5060:
ACK sip:1800373341160@my.domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK59b1364e;rport
Max-Forwards: 70
From: “UTSTAR1” sip:7870@192.168.1.50;tag=as54f61e8c
To: sip:1800373341160@my.domain.com;tag=as2f73c1dd
Contact: sip:7870@192.168.1.50:5060
Call-ID: 669b2260338a832320f155eb6120cbf2@192.168.1.50:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Content-Length: 0

Scheduling destruction of SIP dialog ‘669b2260338a832320f155eb6120cbf2@192.168.1.50:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/5901-0000000d’ status is ‘CHANUNAVAIL’
Really destroying S

6

After the 401 the INVITE is sent with the correct Authorization , then final reponse from YoMaMa server is SIP/2.0 404 Not Found, are you sure you re dialing in th correct format have you verify with them if is needed some kind of prefix

7

figured out my errors

Among other things:
exten => 411,1,Dial(SIP/1800373341160@7870,1)

should be
exten => 411,1,Dial(SIP/18003733411@7870,1)

how the “60” got there is beyond me.

8

That’s why you got the SIP/2.0 404 Not Found,