FXO + POTS + ASTERISK problem when calling out

Asterisk Asterisk Support
1

Hi all. I’m new here.
I tried to find something like my problem but I haven’t found nothing.

I’ve a Asterisk based PBX and It’s linked to the POTS (a standard analogic line) thanks to a gateway FXO.

The problem is that the calls generated from a SIP account logged on the PBX, are not forwarded.

Looking on console: the call stops when
– Attempting native bridge of SIP/815-b6f25858 and SIP/1003-09bd6aa0

Someone could help my to find where is the problem?

Thank you

I’ve enabled the sip debug and copied this code:

– Attempting native bridge of SIP/815-b6f25858 and SIP/1003-09bd6aa0
voismart*CLI>
<-- SIP read from 88.37.114.218:12138:
ACK sip:xxxxxxxxxx@xxxxxxxxxxxx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.232:47860;branch=z9hG4bK-d87543-00224b206e0ec67b-1–d87543-;rport
Max-Forwards: 70
Contact: sip:815@192.168.0.232:47860
To: "xxxxxxxxxxxx"sip:xxxxxxxxxx@xxxxxxxxxxxxx;tag=as145c0796
From: "815"sip:815@xxxxxxxxxxxxxx;tag=7237d056
Call-ID: MDRkMzg1MTRhMGVjNDg5ZGIwNTJhYTgzOGE1MTBmMWM.
CSeq: 2 ACK
Proxy-Authorization: Digest username=“815”,realm=“asterisk”,nonce=“7c139e64”,uri=“sip:xxxxxxxxxxx@xxxxxxxxxxxx”,response=“d61cd860ec66e82cea2b2d4fdf58e945”,algorithm=MD5
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0

— (8 headers 0 lines) —
Destroying call '0ccaa63d08de44a11d3a4c372f9ef2b0@192.168.0.250’
voismart*CLI>
<-- SIP read from 192.168.0.121:5060:
REGISTER sip:192.168.0.250 SIP/2.0
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618
Max-Forwards: 70
Supported: replaces
User-Agent: FXO_GW
Contact: sip:1003@192.168.0.121:5060;q=0.5
Expires: 60
Content-Length: 0

— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.121 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:1003@192.168.0.250
Content-Length: 0

Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250;tag=as21d5bdf0
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="20c1b094"
Content-Length: 0

Scheduling destruction of call ‘c0a80079-13c4-46e50aa5-624e2c3c-31f5’ in 15000 ms
voismart*CLI>

Scheduling destruction of call ‘c0a80079-13c4-46e50aa5-624e2bdd-3700’ in 15000 ms

<-- SIP read from 192.168.0.121:5060:
REGISTER sip:192.168.0.250 SIP/2.0
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618
Max-Forwards: 70
Supported: replaces
User-Agent: FXO_GW
Contact: sip:1003@192.168.0.121:5060;q=0.5
Expires: 60
Content-Length: 0

— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.121 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:1003@192.168.0.250
Content-Length: 0

Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa5-624e2c41-6618;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250;tag=as21d5bdf0
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="20c1b094"
Content-Length: 0

Scheduling destruction of call ‘c0a80079-13c4-46e50aa5-624e2c9b-6656’ in 15000 ms
voismart*CLI>
<-- SIP read from 192.168.0.121:5060:
REGISTER sip:192.168.0.250 SIP/2.0
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf
Max-Forwards: 70
Supported: replaces
User-Agent: FXO_GW
Contact: sip:1003@192.168.0.121:5060;q=0.5
Expires: 60
Authorization: Digest username=“1003”, realm=“asterisk”, nonce=“20c1b094”, uri=“sip:192.168.0.250”, response="eed13d7785bca8f772e02e0fc12c4c80"
Content-Length: 0

— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.121 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:1003@192.168.0.250
Content-Length: 0

Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250;tag=as21d5bdf0
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: sip:1003@192.168.0.121:5060;expires=60
Date: Mon, 10 Sep 2007 09:13:14 GMT
Content-Length: 0

Scheduling destruction of call ‘c0a80079-13c4-46e50aa5-624e2bdd-3700’ in 15000 ms
voismart*CLI>
<-- SIP read from 192.168.0.121:5060:
REGISTER sip:192.168.0.250 SIP/2.0
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf
Max-Forwards: 70
Supported: replaces
User-Agent: FXO_GW
Contact: sip:1003@192.168.0.121:5060;q=0.5
Expires: 60
Authorization: Digest username=“1003”, realm=“asterisk”, nonce=“20c1b094”, uri=“sip:192.168.0.250”, response="eed13d7785bca8f772e02e0fc12c4c80"
Content-Length: 0

— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.121 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:1003@192.168.0.250
Content-Length: 0

Transmitting (no NAT) to 192.168.0.121:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK-46e50aa6-624e2f48-69cf;received=192.168.0.121
From: sip:1003@192.168.0.250;tag=c0a80079-13c4-46e50aa5-624e2c3c-75fb
To: sip:1003@192.168.0.250;tag=as21d5bdf0
Call-ID: c0a80079-13c4-46e50aa5-624e2c3c-31f5
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: sip:1003@192.168.0.121:5060;expires=60
Date: Mon, 10 Sep 2007 09:13:15 GMT
Content-Length: 0

Scheduling destruction of call ‘c0a80079-13c4-46e50aa5-624e2c9b-6656’ in 15000 ms
voismart*CLI>
<-- SIP read from 192.168.0.148:5471:

— (0 headers 0 lines) Nat keepalive —
voismart*CLI>
<-- SIP read from xxxxxxxxxxxxx:26484:

— (0 headers 1 lines) —
voismart*CLI>
<-- SIP read from 192.168.0.232:47860:

2

Please post rel avant part of extensions.conf

Also it is a good idea to remove your real IP’s from your log :wink:

3

Thank you. :smile: I’ve done It.

Sorry… I’m noob… what do you mean with:" rel avant part of extensions.conf"?

4

sorry. Meant to write relevant part. Post the what you are using in extensions.conf to make the call.

5

I’ve not sure but I’ve not full access to extensions.conf.

I’ve discovered that the outgoing calls are managed by an AGI.

It’s the same if I report what the console shows with agi debug enabled (during an outgoing call?)

Thank you