PJSIP config with voiptalk.org/Telappliant

Asterisk Asterisk SIP
1

I am trying to migrate my voiptalk.org trunk from chan_sip to PJSIP, I have updated to asterisk 20.3.1 - I have used the phython conversion script, tried many combinations of various options but so far unable to make any outgoing calls. If anyone has this configuration and can share with me I would much appreciate it or any other help would be appreciated. chan_sip and pjsip config to follow shortly as it will not allow me to post them in this post:

2

SIP.conf

[voiptalk]
type=friend
defaultuser=123456789
secret=xxxxxxxxx
fromuser=123456789
host=voiptalk.org
dtmfmode=rfc2833
fromdomain=voiptalk.org
context=default
;insecure=very
insecure=port,invite
externhost = xxxxxx.ddnsgeek.com
outboundproxy:5065=nat.voiptalk.org

3

PJSIP.conf

[transport-udp]
type = transport
protocol = udp
bind=192.168.1.2
external_media_address = xxxxxx.ddnsgeek.com
external_signaling_address = xxxxxx.ddnsgeek.com
local_net = 192.168.1.0/255.255.255.0

[voiptalk]
type = aor
contact = sip:123456789@voiptalk.org

[voiptalk]
type = identify
endpoint = voiptalk
match = voiptalk.org
match = 217.14.138.0/255.255.255.0
match = 77.240.48.0/255.255.255.0

[voiptalk]
type = auth
username = voiptalk
password = xxxxxxxxx

[voiptalk]
type = endpoint
transport = transport-udp
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g729
allow = h264
allow = h263p
outbound_proxy = sip:nat.voiptalk.org
from_user = 123456789
from_domain = voiptalk.org
language = en_GB
auth = voiptalk
outbound_auth = voiptalk
aors = voiptalk
direct_media = no

4

defaultuser doesn’t do anything here, but may be the cause of the bogus user part in the contact address.

Normal outbound proxies requires ;lr

It is unusual to bind to a specific interface.

The outbound proxy line in the sip.conf appears to be invalid (you can’t have “:” ina a parameter name, but if the intent was to specify the port number, that needs to be in the proxy URI.

Type=friend and insecure=port, in the chan_sip configuration are suboptimal.

You seem to have forgotten to provide the logs.

5

Thanks for the reply. sip.conf config works absolutely fine, it is the pjsip.conf config that is causing me a problem.

I have updated pjsip.conf as per the below plus logs but tesult still the same no connection
:
PJSIP.CONF
;–
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
videosupport = yes

[voiptalk]
defaultuser = 123456789

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
–;

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = xxxxxx.ddnsgeek.com
external_signaling_address = xxxxxx.ddnsgeek.com
local_net = 192.168.1.0/255.255.255.0

[voiptalk]
type = aor
contact = sip:xxxxxxxx@voiptalk.org

[voiptalk]
type = identify
endpoint = voiptalk
match = 217.14.138.0/255.255.255.0
match = 77.240.48.0/255.255.255.0

[voiptalk]
type = auth
username = voiptalk
password = ???

[voiptalk]
type = endpoint
transport = transport-udp
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g729
allow = h264
allow = h263p
outbound_proxy = nat.voiptalk.org:5065;lr
from_user = xxxxxxxx
language = en_GB
auth = voiptalk
outbound_auth = voiptalk
aors = voiptalk
direct_media = no

LOGS:
<— Received SIP request (998 bytes) from UDP:192.168.1.85:5060 —>
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2db04f74989634e620d0acef72316d1a;rport
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3299563061@192_168_1_85
CSeq: 2 INVITE
Contact: sip:12@192.168.1.85:5060
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10690 202 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10690 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<— Transmitting SIP response (533 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK2db04f74989634e620d0acef72316d1a
Call-ID: 3299563061@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone;tag=z9hG4bK2db04f74989634e620d0acef72316d1a
CSeq: 2 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1689425510/e64cf64ca8cc19f33ca44831bafaa45d”,opaque=“6a2dbd7220b93713”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Received SIP request (450 bytes) from UDP:192.168.1.85:5060 —>
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2db04f74989634e620d0acef72316d1a;rport
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone;tag=z9hG4bK2db04f74989634e620d0acef72316d1a
Call-ID: 3299563061@192_168_1_85
CSeq: 2 ACK
Contact: sip:12@192.168.1.85:5060
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

<— Received SIP request (1304 bytes) from UDP:192.168.1.85:5060 —>
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3299563061@192_168_1_85
CSeq: 3 INVITE
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689425510/e64cf64ca8cc19f33ca44831bafaa45d”, nc=00000001, cnonce=“af6e47935dd3cb0c2f74bb361224ba89”, opaque=“6a2dbd7220b93713”, response=“b06b14a213c64a033e7f9cbc251a6038”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10690 202 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10690 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<— Transmitting SIP response (337 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Content-Length: 0

-- Executing [8307887090909@internal:1] Dial("PJSIP/12-00000002", "PJSIP/07887090909@voiptalk") in new stack
-- Called PJSIP/07887090909@voiptalk

<— Received SIP request (718 bytes) from UDP:192.168.1.85:5060 —>
CANCEL sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3299563061@192_168_1_85
CSeq: 3 CANCEL
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689425510/e64cf64ca8cc19f33ca44831bafaa45d”, nc=00000002, cnonce=“37add1b3b7ed08b8bab6ac0a905a0734”, opaque=“6a2dbd7220b93713”, response=“fda1e03d1ba67840e78ed9beb8ebaa18”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

<— Transmitting SIP response (374 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
CSeq: 3 CANCEL
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Transmitting SIP response (413 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Reason: Q.850;cause=0
Content-Length: 0

== Spawn extension (internal, 8307887090909, 1) exited non-zero on ‘PJSIP/12-00000002’
<— Received SIP request (753 bytes) from UDP:192.168.1.85:5060 —>
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: “Kitchen <12>” sip:12@asterisk;tag=2391507181
To: sip:8307887090909@asterisk;user=phone;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
Call-ID: 3299563061@192_168_1_85
CSeq: 3 ACK
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689425510/e64cf64ca8cc19f33ca44831bafaa45d”, nc=00000001, cnonce=“af6e47935dd3cb0c2f74bb361224ba89”, opaque=“6a2dbd7220b93713”, response=“b06b14a213c64a033e7f9cbc251a6038”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

6

When you are using chan_sip, does voiptalk really accept a Caller ID of
“Kitchen <12>” ?

That surprises me.

Antony.

7

:
PJSIP.CONF

;--
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
videosupport = yes

[voiptalk]
defaultuser = 123456789


;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
--;

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = banyan.ddnsgeek.com
external_signaling_address = banyan.ddnsgeek.com
local_net = 192.168.1.0/255.255.255.0


[voiptalk]
type = aor
contact = sip:xxxxxxxx@voiptalk.org

[voiptalk]
type = identify
endpoint = voiptalk
match = 217.14.138.0/255.255.255.0
match = 77.240.48.0/255.255.255.0


[voiptalk]
type = auth
username = voiptalk
password = ???????

[voiptalk]
type = endpoint
transport = transport-udp
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g729
allow = h264
allow = h263p
outbound_proxy = nat.voiptalk.org:5065\;lr
from_user = xxxxxxxx
language = en_GB
auth = voiptalk
outbound_auth = voiptalk
aors = voiptalk
direct_media = no

LOGS:

<--- Received SIP request (998 bytes) from UDP:192.168.1.85:5060 --->
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2db04f74989634e620d0acef72316d1a;rport
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>
Call-ID: 3299563061@192_168_1_85
CSeq: 2 INVITE
Contact: <sip:12@192.168.1.85:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10690 202 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10690 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<--- Transmitting SIP response (533 bytes) to UDP:192.168.1.85:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK2db04f74989634e620d0acef72316d1a
Call-ID: 3299563061@192_168_1_85
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>;tag=z9hG4bK2db04f74989634e620d0acef72316d1a
CSeq: 2 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1689425510/e64cf64ca8cc19f33ca44831bafaa45d",opaque="6a2dbd7220b93713",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.3.1
Content-Length:  0


<--- Received SIP request (450 bytes) from UDP:192.168.1.85:5060 --->
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK2db04f74989634e620d0acef72316d1a;rport
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>;tag=z9hG4bK2db04f74989634e620d0acef72316d1a
Call-ID: 3299563061@192_168_1_85
CSeq: 2 ACK
Contact: <sip:12@192.168.1.85:5060>
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0


<--- Received SIP request (1304 bytes) from UDP:192.168.1.85:5060 --->
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>
Call-ID: 3299563061@192_168_1_85
CSeq: 3 INVITE
Contact: <sip:12@192.168.1.85:5060>
Authorization: Digest username="12", realm="asterisk", qop=auth, algorithm=MD5, uri="sip:8307887090909@asterisk;user=phone", nonce="1689425510/e64cf64ca8cc19f33ca44831bafaa45d", nc=00000001, cnonce="af6e47935dd3cb0c2f74bb361224ba89", opaque="6a2dbd7220b93713", response="b06b14a213c64a033e7f9cbc251a6038"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10690 202 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10690 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<--- Transmitting SIP response (337 bytes) to UDP:192.168.1.85:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Content-Length:  0


    -- Executing [8307887090909@internal:1] Dial("PJSIP/12-00000002", "PJSIP/07887090909@voiptalk") in new stack
    -- Called PJSIP/07887090909@voiptalk
<--- Received SIP request (718 bytes) from UDP:192.168.1.85:5060 --->
CANCEL sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>
Call-ID: 3299563061@192_168_1_85
CSeq: 3 CANCEL
Contact: <sip:12@192.168.1.85:5060>
Authorization: Digest username="12", realm="asterisk", qop=auth, algorithm=MD5, uri="sip:8307887090909@asterisk;user=phone", nonce="1689425510/e64cf64ca8cc19f33ca44831bafaa45d", nc=00000002, cnonce="37add1b3b7ed08b8bab6ac0a905a0734", opaque="6a2dbd7220b93713", response="fda1e03d1ba67840e78ed9beb8ebaa18"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0


<--- Transmitting SIP response (374 bytes) to UDP:192.168.1.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
CSeq: 3 CANCEL
Server: Asterisk PBX 20.3.1
Content-Length:  0


<--- Transmitting SIP response (413 bytes) to UDP:192.168.1.85:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba
Call-ID: 3299563061@192_168_1_85
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Reason: Q.850;cause=0
Content-Length:  0


  == Spawn extension (internal, 8307887090909, 1) exited non-zero on 'PJSIP/12-00000002'
<--- Received SIP request (753 bytes) from UDP:192.168.1.85:5060 --->
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK29a8f1a916320ec8c7dbefb4532738ba;rport
From: "Kitchen <12>" <sip:12@asterisk>;tag=2391507181
To: <sip:8307887090909@asterisk;user=phone>;tag=0dc15bd7-951f-430b-817d-6c2f4cc4fd13
Call-ID: 3299563061@192_168_1_85
CSeq: 3 ACK
Contact: <sip:12@192.168.1.85:5060>
Authorization: Digest username="12", realm="asterisk", qop=auth, algorithm=MD5, uri="sip:8307887090909@asterisk;user=phone", nonce="1689425510/e64cf64ca8cc19f33ca44831bafaa45d", nc=00000001, cnonce="af6e47935dd3cb0c2f74bb361224ba89", opaque="6a2dbd7220b93713", response="b06b14a213c64a033e7f9cbc251a6038"
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

This should be removed, and possibly replaced by hand translated equivalents.

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements start
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[general]
videosupport = yes

[voiptalk]
defaultuser = 123456789


;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Non mapped elements end
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
--;

This shouldn’t have a user part (the called number goes there):

contact = sip:xxxxxxxx@voiptalk.org

Your logs show neither outbound traffic nor an error. I wonder if you have a DNS problem.

PJSIP.CONF should be pjsip.conf

8

OK I have made those modifications and tested again but still no success. My asterisk switch is behind a NAT firewall but have had no other DNS problems. Here is the updated pjsip and logs:

pjsip.conf

[general]
videosupport = yes

[voiptalk]
defaultuser = xxxxxxxx

[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = xxxxxxxx.ddnsgeek.com
external_signaling_address = xxxxxxxx.ddnsgeek.com
local_net = 192.168.1.0/255.255.255.0

[voiptalk]
type = aor
contact = sip:voiptalk.org:5060

[voiptalk]
type = identify
endpoint = voiptalk
match = 217.14.138.0/255.255.255.0
match = 77.240.48.0/255.255.255.0

[voiptalk]
type = auth
username = voiptalk
password = ???

[voiptalk]
type = endpoint
transport = transport-udp
context = default
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
allow = gsm
allow = g729
allow = h264
allow = h263p
outbound_proxy = nat.voiptalk.org:5065;lr
from_user = xxxxxxxx
from_domain = voiptalk.org
language = en_GB
auth = voiptalk
outbound_auth = voiptalk
aors = voiptalk
force_rport = yes
direct_media = no

LOGS
PJSIP Logging enabled
<— Received SIP request (998 bytes) from UDP:192.168.1.85:5060 —>
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK9970867642b1c5db96c0a8a5db5cb948;rport
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3410868825@192_168_1_85
CSeq: 2 INVITE
Contact: sip:12@192.168.1.85:5060
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10694 204 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10694 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<— Transmitting SIP response (533 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK9970867642b1c5db96c0a8a5db5cb948
Call-ID: 3410868825@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone;tag=z9hG4bK9970867642b1c5db96c0a8a5db5cb948
CSeq: 2 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1689429303/7b6f9f7ef91cf7d7be912667669c2ce6”,opaque=“3fb85aaa2a3fa5a3”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Received SIP request (450 bytes) from UDP:192.168.1.85:5060 —>
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK9970867642b1c5db96c0a8a5db5cb948;rport
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone;tag=z9hG4bK9970867642b1c5db96c0a8a5db5cb948
Call-ID: 3410868825@192_168_1_85
CSeq: 2 ACK
Contact: sip:12@192.168.1.85:5060
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

<— Received SIP request (1303 bytes) from UDP:192.168.1.85:5060 —>
INVITE sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0;rport
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3410868825@192_168_1_85
CSeq: 3 INVITE
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689429303/7b6f9f7ef91cf7d7be912667669c2ce6”, nc=00000001, cnonce=“1aa3f05c59fd03363f808445fc8c506”, opaque=“3fb85aaa2a3fa5a3”, response=“c1ce30ac310b9f306b7f0c64b8588ef3”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 382

v=0
o=12 10694 204 IN IP4 192.168.1.85
s=Mapping
c=IN IP4 192.168.1.85
t=0 0
m=audio 10694 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<— Transmitting SIP response (337 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0
Call-ID: 3410868825@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Content-Length: 0

-- Executing [8307887090909@internal:1] Dial("PJSIP/12-00000002", "PJSIP/07887090909@voiptalk") in new stack
-- Called PJSIP/07887090909@voiptalk

<— Received SIP request (476 bytes) from UDP:192.168.1.81:5060 —>
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.81:5060;branch=z9hG4bKf899612eb89ec5529131ee3c4ac67cbe;rport
From: “Guest 2nd Fl” sip:34@asterisk;tag=2739580058
To: “Guest 2nd Fl” sip:34@asterisk
Call-ID: 97022773@192_168_1_81
CSeq: 700 REGISTER
Contact: sip:34@192.168.1.81:5060
Max-Forwards: 70
User-Agent: N300 IP/42.262.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0

<— Transmitting SIP response (528 bytes) to UDP:192.168.1.81:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.81:5060;rport=5060;received=192.168.1.81;branch=z9hG4bKf899612eb89ec5529131ee3c4ac67cbe
Call-ID: 97022773@192_168_1_81
From: “Guest 2nd Fl” sip:34@asterisk;tag=2739580058
To: “Guest 2nd Fl” sip:34@asterisk;tag=z9hG4bKf899612eb89ec5529131ee3c4ac67cbe
CSeq: 700 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1689429311/aa8563e47b1b0b41181d2b54fc2910cc”,opaque=“731953d719a5af3a”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Received SIP request (756 bytes) from UDP:192.168.1.81:5060 —>
REGISTER sip:asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.81:5060;branch=z9hG4bKb3f9058950904565a6e014b8be5036ad;rport
From: “Guest 2nd Fl” sip:34@asterisk;tag=2739580058
To: “Guest 2nd Fl” sip:34@asterisk
Call-ID: 97022773@192_168_1_81
CSeq: 701 REGISTER
Contact: sip:34@192.168.1.81:5060
Authorization: Digest username=“34”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:asterisk”, nonce=“1689429311/aa8563e47b1b0b41181d2b54fc2910cc”, nc=00000001, cnonce=“19b21366ab822e7711844e8510f57e6”, opaque=“731953d719a5af3a”, response=“b2c573678f338da90b20ccbf5c51cdce”
Max-Forwards: 70
User-Agent: N300 IP/42.262.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Length: 0

<— Transmitting SIP response (472 bytes) to UDP:192.168.1.81:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.81:5060;rport=5060;received=192.168.1.81;branch=z9hG4bKb3f9058950904565a6e014b8be5036ad
Call-ID: 97022773@192_168_1_81
From: “Guest 2nd Fl” sip:34@asterisk;tag=2739580058
To: “Guest 2nd Fl” sip:34@asterisk;tag=z9hG4bKb3f9058950904565a6e014b8be5036ad
CSeq: 701 REGISTER
Date: Sat, 15 Jul 2023 13:55:11 GMT
Contact: sip:34@192.168.1.81:5060;expires=179
Expires: 180
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Received SIP request (718 bytes) from UDP:192.168.1.85:5060 —>
CANCEL sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0;rport
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone
Call-ID: 3410868825@192_168_1_85
CSeq: 3 CANCEL
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689429303/7b6f9f7ef91cf7d7be912667669c2ce6”, nc=00000002, cnonce=“8fc22f399f06e5fc36b702ad4accd9e9”, opaque=“3fb85aaa2a3fa5a3”, response=“0f73c8bd178abb1d42e32753d80b03b6”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

<— Transmitting SIP response (374 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0
Call-ID: 3410868825@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone;tag=6a9a603f-4a07-464d-83ec-4c3cfb927704
CSeq: 3 CANCEL
Server: Asterisk PBX 20.3.1
Content-Length: 0

<— Transmitting SIP response (413 bytes) to UDP:192.168.1.85:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.85:5060;rport=5060;received=192.168.1.85;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0
Call-ID: 3410868825@192_168_1_85
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone;tag=6a9a603f-4a07-464d-83ec-4c3cfb927704
CSeq: 3 INVITE
Server: Asterisk PBX 20.3.1
Reason: Q.850;cause=0
Content-Length: 0

== Spawn extension (internal, 8307887090909, 1) exited non-zero on ‘PJSIP/12-00000002’
<— Received SIP request (752 bytes) from UDP:192.168.1.85:5060 —>
ACK sip:8307887090909@asterisk;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.85:5060;branch=z9hG4bK897bad3ad091c31e4fa8392bd2438fa0;rport
From: “Kitchen <12>” sip:12@asterisk;tag=1184633702
To: sip:8307887090909@asterisk;user=phone;tag=6a9a603f-4a07-464d-83ec-4c3cfb927704
Call-ID: 3410868825@192_168_1_85
CSeq: 3 ACK
Contact: sip:12@192.168.1.85:5060
Authorization: Digest username=“12”, realm=“asterisk”, qop=auth, algorithm=MD5, uri=“sip:8307887090909@asterisk;user=phone”, nonce=“1689429303/7b6f9f7ef91cf7d7be912667669c2ce6”, nc=00000001, cnonce=“1aa3f05c59fd03363f808445fc8c506”, opaque=“3fb85aaa2a3fa5a3”, response=“c1ce30ac310b9f306b7f0c64b8588ef3”
Max-Forwards: 70
User-Agent: N510 IP PRO/42.263.00.000.000
Content-Length: 0

9

Could you please markup your logs and configuration as pre-formatted text, as I did.

Also please don’t screen scrape logs, as you lose important timing information; use the log files.

You appear to have turned the verbosity down, as I can’t see the dialplan execution at all this time. You probably actually need to turn it up, as the problem was it was showing as calling Dial(), then nothing happened.

This is what was there last time, but not here this time. It should have resulted in either a failure, or showing an outgoing INVITE:

    -- Executing [8307887090909@internal:1] Dial("PJSIP/12-00000002", "PJSIP/07887090909@voiptalk") in new stack
    -- Called PJSIP/07887090909@voiptalk
10 
outbound_proxy = nat.voiptalk.org:5065;lr

This is not valid. It should be:

outbound_proxy = sip:nat.voiptalk.org:5065\;lr
11

Oops missed that. The OP had the scheme part right originally and I didn’t look to see if that got broken.

12

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