Can't Make Outbound Calls Using PJSIP

moon-fox · May 21, 2023, 3:30am

I’m using PJSIP. I can register to the PSTN and able to accept incoming calls but I cannot make outbound calls.

This is how my pisjip.conf looks like:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[023545090]
type=registration
outbound_auth=023545090
server_uri=sip:sipapvpbx.totbb.net
client_uri=sip:023545090@sipapvpbx.totbb.net
retry_interval=60
[023545090]
type=auth
auth_type=userpass
password=xxxxxxxxx
username=023545090
[023545090]
type=aor
contact=sip:sipapvpbx.totbb.net
[023545090]
type=endpoint
context=incoming
disallow=all
allow=ulaw
outbound_auth=023545090
aors=023545090
from_user=023545090
from_domain=sipapvpbx.totbb.net
[023545090]
type=identify
endpoint=023545090
match=sipapvpbx.totbb.net

and here is the logs

<--- Received SIP request (1053 bytes) from UDP:10.0.5.252:51710 --->
INVITE sip:023545096@10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPj02U9Xe5x0Gv9HSbV419U97LfuO3VT8zS
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: sip:023545096@10.0.4.10
Contact: "6001" <sip:6001@10.0.5.252:51710;ob>
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
CSeq: 28726 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Content-Type: application/sdp
Content-Length:   471

v=0
o=- 3893628269 3893628269 IN IP4 10.0.5.252
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4016 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.5.252
b=TIAS:96000
a=rtcp:4017 IN IP4 10.0.5.252
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:1402785162 cname:4271c8193c080363

<--- Transmitting SIP response (552 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPj02U9Xe5x0Gv9HSbV419U97LfuO3VT8zS
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: <sip:023545096@10.0.4.10>;tag=z9hG4bKPj02U9Xe5x0Gv9HSbV419U97LfuO3VT8zS
CSeq: 28726 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1684639469/30e26c1c5daf1b22f66ee9cb59aefdb6",opaque="666010951ce5a0fd",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.17.1
Content-Length:  0


<--- Received SIP request (374 bytes) from UDP:10.0.5.252:51710 --->
ACK sip:023545096@10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPj02U9Xe5x0Gv9HSbV419U97LfuO3VT8zS
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: sip:023545096@10.0.4.10;tag=z9hG4bKPj02U9Xe5x0Gv9HSbV419U97LfuO3VT8zS
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
CSeq: 28726 ACK
Content-Length:  0


<--- Received SIP request (1346 bytes) from UDP:10.0.5.252:51710 --->
INVITE sip:023545096@10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPjLUrsWoIYf0516.XHiRUBJBPxSpSRgyYE
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: sip:023545096@10.0.4.10
Contact: "6001" <sip:6001@10.0.5.252:51710;ob>
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
CSeq: 28727 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.6
Authorization: Digest username="6001", realm="asterisk", nonce="1684639469/30e26c1c5daf1b22f66ee9cb59aefdb6", uri="sip:023545096@10.0.4.10", response="812462caa09524066bebb71d646c9b8a", algorithm=MD5, cnonce="Uk22SY0W0OqAywMR4TGq22vomZSOSJE", opaque="666010951ce5a0fd", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   471

v=0
o=- 3893628269 3893628269 IN IP4 10.0.5.252
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 4016 RTP/AVP 96 9 8 0 101 102
c=IN IP4 10.0.5.252
b=TIAS:96000
a=rtcp:4017 IN IP4 10.0.5.252
a=sendrecv
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ssrc:1402785162 cname:4271c8193c080363

<--- Transmitting SIP response (354 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPjLUrsWoIYf0516.XHiRUBJBPxSpSRgyYE
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: <sip:023545096@10.0.4.10>
CSeq: 28727 INVITE
Server: Asterisk PBX 18.17.1
Content-Length:  0


    -- Executing [023545096@outgoing:1] Set("PJSIP/6001-00000004", "CALLERID(num)=023545090") in new stack
    -- Executing [023545096@outgoing:2] Dial("PJSIP/6001-00000004", "PJSIP/023545096@023545090") in new stack
    -- Called PJSIP/023545096@023545090
<--- Transmitting SIP request (918 bytes) to UDP:182.53.221.118:5060 --->
INVITE sip:023545096@sipapvpbx.totbb.net SIP/2.0
Via: SIP/2.0/UDP 10.0.4.10:5060;rport;branch=z9hG4bKPj7285d9e6-48e5-4df1-a795-9adf06c3c175
From: <sip:023545090@sipapvpbx.totbb.net>;tag=3e188806-c4b4-4dcc-af1a-9ac9709f627b
To: <sip:023545096@sipapvpbx.totbb.net>
Contact: <sip:023545090@10.0.4.10:5060>
Call-ID: 61405753-493c-4e1e-81b1-24f614e95470
CSeq: 24678 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.17.1
Content-Type: application/sdp
Content-Length:   231

v=0
o=- 1442236409 1442236409 IN IP4 10.0.4.10
s=Asterisk
c=IN IP4 10.0.4.10
t=0 0
m=audio 12558 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 response (399 bytes) from UDP:182.53.221.118:5060 --->
SIP/2.0 100 Trying
Call-ID: 61405753-493c-4e1e-81b1-24f614e95470
CSeq: 24678 INVITE
From: <sip:023545090@sipapvpbx.totbb.net>;tag=3e188806-c4b4-4dcc-af1a-9ac9709f627b
To: <sip:023545096@sipapvpbx.totbb.net>;tag=sip+3+ef4e0073+7873171e
Via: SIP/2.0/UDP 10.0.4.10:5060;received=182.52.129.135;rport=5060;branch=z9hG4bKPj7285d9e6-48e5-4df1-a795-9adf06c3c175
Server: SIP/2.0
Content-Length: 0


<--- Received SIP response (476 bytes) from UDP:182.53.221.118:5060 --->
SIP/2.0 403 Forbidden
Call-ID: 61405753-493c-4e1e-81b1-24f614e95470
CSeq: 24678 INVITE
From: <sip:023545090@sipapvpbx.totbb.net>;tag=3e188806-c4b4-4dcc-af1a-9ac9709f627b
To: <sip:023545096@sipapvpbx.totbb.net>;tag=sip+3+ef4e0073+7873171e
Via: SIP/2.0/UDP 10.0.4.10:5060;received=182.52.129.135;rport=5060;branch=z9hG4bKPj7285d9e6-48e5-4df1-a795-9adf06c3c175
Content-Length: 0
Contact: <sip:182.53.221.118:5060>
Server: DC-SIP/2.0
Organization: Metaswitch Networks


<--- Transmitting SIP request (430 bytes) to UDP:182.53.221.118:5060 --->
ACK sip:023545096@sipapvpbx.totbb.net SIP/2.0
Via: SIP/2.0/UDP 10.0.4.10:5060;rport;branch=z9hG4bKPj7285d9e6-48e5-4df1-a795-9adf06c3c175
From: <sip:023545090@sipapvpbx.totbb.net>;tag=3e188806-c4b4-4dcc-af1a-9ac9709f627b
To: <sip:023545096@sipapvpbx.totbb.net>;tag=sip+3+ef4e0073+7873171e
Call-ID: 61405753-493c-4e1e-81b1-24f614e95470
CSeq: 24678 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.17.1
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/6001-00000004' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (432 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPjLUrsWoIYf0516.XHiRUBJBPxSpSRgyYE
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: <sip:023545096@10.0.4.10>;tag=0a772575-63e9-448a-8f8d-4a8167e92f0f
CSeq: 28727 INVITE
Server: Asterisk PBX 18.17.1
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (369 bytes) from UDP:10.0.5.252:51710 --->
ACK sip:023545096@10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPjLUrsWoIYf0516.XHiRUBJBPxSpSRgyYE
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=PGWly9u-dz0pvdr8DQIs-0N-6nLad.xd
To: sip:023545096@10.0.4.10;tag=0a772575-63e9-448a-8f8d-4a8167e92f0f
Call-ID: 5VbYRXghWHxCMzCFhU29ZN6kTfY6JqcI
CSeq: 28727 ACK
Content-Length:  0


<--- Received SIP request (518 bytes) from UDP:10.0.5.252:51710 --->
REGISTER sip:10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPjUYPIUjktUiIyeYV9QEGlxi6hdZpg3YDP
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=QUQNuEazZaoiFA8e7WquKdoezaIjLQJU
To: "6001" <sip:6001@10.0.4.10>
Call-ID: KhHCilROcuW3aa10PqhuER8BwfDL5fvR
CSeq: 5626 REGISTER
User-Agent: Telephone 1.6
Contact: "6001" <sip:6001@10.0.5.252:51710;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<--- Transmitting SIP response (555 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPjUYPIUjktUiIyeYV9QEGlxi6hdZpg3YDP
Call-ID: KhHCilROcuW3aa10PqhuER8BwfDL5fvR
From: "6001" <sip:6001@10.0.4.10>;tag=QUQNuEazZaoiFA8e7WquKdoezaIjLQJU
To: "6001" <sip:6001@10.0.4.10>;tag=z9hG4bKPjUYPIUjktUiIyeYV9QEGlxi6hdZpg3YDP
CSeq: 5626 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1684639676/434e5e196167b052c4d08ccf9afd19e1",opaque="4b4ef3d133e7ca7e",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.17.1
Content-Length:  0


<--- Received SIP request (801 bytes) from UDP:10.0.5.252:51710 --->
REGISTER sip:10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPj5ffySjp9DWN6uvldGYBrs4jzhtnhJm2t
Max-Forwards: 70
From: "6001" <sip:6001@10.0.4.10>;tag=QUQNuEazZaoiFA8e7WquKdoezaIjLQJU
To: "6001" <sip:6001@10.0.4.10>
Call-ID: KhHCilROcuW3aa10PqhuER8BwfDL5fvR
CSeq: 5627 REGISTER
User-Agent: Telephone 1.6
Contact: "6001" <sip:6001@10.0.5.252:51710;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="6001", realm="asterisk", nonce="1684639676/434e5e196167b052c4d08ccf9afd19e1", uri="sip:10.0.4.10", response="f022a74469c2b187b63d46b339dd31a0", algorithm=MD5, cnonce="WcA0YJeP2.hk.P1JYwxY.ytnqdP8eeQ", opaque="4b4ef3d133e7ca7e", qop=auth, nc=00000001
Content-Length:  0


<--- Transmitting SIP response (503 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPj5ffySjp9DWN6uvldGYBrs4jzhtnhJm2t
Call-ID: KhHCilROcuW3aa10PqhuER8BwfDL5fvR
From: "6001" <sip:6001@10.0.4.10>;tag=QUQNuEazZaoiFA8e7WquKdoezaIjLQJU
To: "6001" <sip:6001@10.0.4.10>;tag=z9hG4bKPj5ffySjp9DWN6uvldGYBrs4jzhtnhJm2t
CSeq: 5627 REGISTER
Date: Sun, 21 May 2023 03:27:56 GMT
Contact: <sip:6001@10.0.5.252:51710;ob>;expires=299
Expires: 300
Server: Asterisk PBX 18.17.1
Content-Length:  0


<--- Received SIP request (519 bytes) from UDP:10.0.5.252:51710 --->
REGISTER sip:10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPjwOph-qi-2BK5nVSwP-O3EhKU0Vtb0r8T
Max-Forwards: 70
From: "6002" <sip:6002@10.0.4.10>;tag=AwaB-o0vKVK9IQh99Yy6db-X2gswHwhK
To: "6002" <sip:6002@10.0.4.10>
Call-ID: UQBORVCY6NzB7pG8b85YzO44gqlasCQ2
CSeq: 59958 REGISTER
User-Agent: Telephone 1.6
Contact: "6002" <sip:6002@10.0.5.252:51710;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<--- Transmitting SIP response (556 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPjwOph-qi-2BK5nVSwP-O3EhKU0Vtb0r8T
Call-ID: UQBORVCY6NzB7pG8b85YzO44gqlasCQ2
From: "6002" <sip:6002@10.0.4.10>;tag=AwaB-o0vKVK9IQh99Yy6db-X2gswHwhK
To: "6002" <sip:6002@10.0.4.10>;tag=z9hG4bKPjwOph-qi-2BK5nVSwP-O3EhKU0Vtb0r8T
CSeq: 59958 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1684639704/48cbd74a2a7b1136ecebac9810bec87f",opaque="7696b24628b1d859",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.17.1
Content-Length:  0


<--- Received SIP request (802 bytes) from UDP:10.0.5.252:51710 --->
REGISTER sip:10.0.4.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.252:51710;rport;branch=z9hG4bKPjoaIl2.40ej1IH3xqA-Mg-RreN8zRWbxI
Max-Forwards: 70
From: "6002" <sip:6002@10.0.4.10>;tag=AwaB-o0vKVK9IQh99Yy6db-X2gswHwhK
To: "6002" <sip:6002@10.0.4.10>
Call-ID: UQBORVCY6NzB7pG8b85YzO44gqlasCQ2
CSeq: 59959 REGISTER
User-Agent: Telephone 1.6
Contact: "6002" <sip:6002@10.0.5.252:51710;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="6002", realm="asterisk", nonce="1684639704/48cbd74a2a7b1136ecebac9810bec87f", uri="sip:10.0.4.10", response="579b55c4b0a84f276184301e05fbcc8d", algorithm=MD5, cnonce="DZVc3sz687m26fj5qhy0o1PMUL4EyHf", opaque="7696b24628b1d859", qop=auth, nc=00000001
Content-Length:  0


<--- Transmitting SIP response (504 bytes) to UDP:10.0.5.252:51710 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.5.252:51710;rport=51710;received=10.0.5.252;branch=z9hG4bKPjoaIl2.40ej1IH3xqA-Mg-RreN8zRWbxI
Call-ID: UQBORVCY6NzB7pG8b85YzO44gqlasCQ2
From: "6002" <sip:6002@10.0.4.10>;tag=AwaB-o0vKVK9IQh99Yy6db-X2gswHwhK
To: "6002" <sip:6002@10.0.4.10>;tag=z9hG4bKPjoaIl2.40ej1IH3xqA-Mg-RreN8zRWbxI
CSeq: 59959 REGISTER
Date: Sun, 21 May 2023 03:28:24 GMT
Contact: <sip:6002@10.0.5.252:51710;ob>;expires=299
Expires: 300
Server: Asterisk PBX 18.17.1
Content-Length:  0

it gives this error when I try to make a call

    -- Executing [023545096@outgoing:1] Set("PJSIP/6001-00000008", "CALLERID(num)=023545090") in new stack
    -- Executing [023545096@outgoing:2] Dial("PJSIP/6001-00000008", "PJSIP/023545096@023545090") in new stack
    -- Called PJSIP/023545096@023545090
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/6001-00000008' status is 'CHANUNAVAIL'

jcolp · May 21, 2023, 8:56am

Whatever you have called responded with a “SIP/2.0 403 Forbidden” and rejected the call, so it’s likely configuration. From an Asterisk side the configuration makes sense, so you’ll probably have to engage with the remote side to determine why it is responding as such.

Pooh · May 21, 2023, 9:07am

So, your provider sent you a SIP 403 Forbidden, and then a 503 Service
Unavailable, with a Q.850 reason code of 34, which means “No circuit/channel
available”, which generally indicates that there is no appropriate
circuit/channel presently available to handle the call.

In this situation I would ask your SIP provider:

whether you have outbound dialling capability on your service
whether the format of the number you are dialling out to is correct
whether the format of your Caller ID is correct
whether they have any other requirements (such as PAID header) which you
need to supply in order to be able to dial out

Antony.

david551 · May 21, 2023, 11:35am

He didn’t receive this; he sent it. And 503 is pretty much the catch all used by Asterisk when it doesn’t have a better response. I think 503 for 403 has been discussed before.

rvk · May 22, 2023, 3:35pm

Is that 10.0.4.10 your internal IP for the Asterisk server?
Shouldn’t it pass the external IP in the invite when placing a call?

Did you set the external_media_address and external_signaling_address in your pjsip.conf correctly?

system · June 21, 2023, 3:48pm

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Topic		Replies	Views
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Can't Make Outbound Calls Using PJSIP

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