Trunk PSTN - dont work 503

Asterisk Asterisk SIP
1

Dear,

We have problems with a Telecom Arg trunk. It is a trunk that based on IP authentication which goes without user or password authentication, uses a static route to a provider gateway and at the same time a second network card is configured in the central office. with an ip that the provider gives us and a gateway that is the one that allows us to reach the sip gateway but we have retransmissions of this trunk all the time.

What can be the drawback? We have already tried several configurations on the trunk side and none of them works, pinging gateways for both the subnet of the second network card and the sip gateway are fine.

Telecom Arg. Says that the problem is a matter of commutation of our pbx. Since you see “401 UNAUTHORIZED”. But we only view the first 401, but de conmutations continues. the PSTN never responde.

Peer details:

PEER Details:

host=190.224.166.135
type=peer
dtmfmode=rfc2833
disallow=all
allow=g729&alaw&ulaw

2020/05/21 11:56:39.524871 172.27.160.47:5060 -> 192.168.20.15:5060
INVITE sip:51991206@192.168.20.15 SIP/2.0
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-a82a630f6d6c4e157f5cf9e9c2f57db8
Max-Forwards: 70
Contact: "501" <sip:501@172.27.160.47:5060;transport=udp;registering_acc=192_168_20_15>
User-Agent: Jitsi2.10.5550Linux
Content-Type: application/sdp
Content-Length: 289

v=0
o=501-jitsi.org 0 0 IN IP4 172.27.160.47
s=-
c=IN IP4 172.27.160.47
t=0 0
m=audio 5113 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics


2020/05/21 11:56:39.525299 192.168.20.15:5060 -> 172.27.160.47:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-a82a630f6d6c4e157f5cf9e9c2f57db8;received=172.27.160.47
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>;tag=as235dbfe2
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 1 INVITE
Server: IPBX-2.11.0.47(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f07e7cb"
Content-Length: 0



2020/05/21 11:56:39.550520 172.27.160.47:5060 -> 192.168.20.15:5060
ACK sip:51991206@192.168.20.15 SIP/2.0
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>;tag=as235dbfe2
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-a82a630f6d6c4e157f5cf9e9c2f57db8
CSeq: 1 ACK
Content-Length: 0



2020/05/21 11:56:39.550534 172.27.160.47:5060 -> 192.168.20.15:5060
INVITE sip:51991206@192.168.20.15 SIP/2.0
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>
Max-Forwards: 70
Contact: "501" <sip:501@172.27.160.47:5060;transport=udp;registering_acc=192_168_20_15>
User-Agent: Jitsi2.10.5550Linux
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-6bc0bd877ffd6e88ee83492b0a9c2bb5
Authorization: Digest username="501",realm="asterisk",nonce="6f07e7cb",uri="sip:51991206@192.168.20.15",response="64b9e1e03090c3e3ec1a63c90d939d83",algorithm=MD5
Content-Length: 289

v=0
o=501-jitsi.org 0 0 IN IP4 172.27.160.47
s=-
c=IN IP4 172.27.160.47
t=0 0
m=audio 5113 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-xr:voip-metrics


2020/05/21 11:56:39.552392 192.168.20.15:5060 -> 172.27.160.47:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-6bc0bd877ffd6e88ee83492b0a9c2bb5;received=172.27.160.47
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: IPBX-2.11.0.47(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:51991206@192.168.20.15:5060>
Content-Length: 0



2020/05/21 11:57:11.860952 192.168.20.15:5060 -> 172.27.160.47:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-6bc0bd877ffd6e88ee83492b0a9c2bb5;received=172.27.160.47
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>;tag=as0f379032
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: IPBX-2.11.0.47(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:51991206@192.168.20.15:5060>
Content-Type: application/sdp
Content-Length: 206

v=0
o=root 1926549177 1926549177 IN IP4 192.168.20.15
s=Asterisk PBX 11.25.3
c=IN IP4 192.168.20.15
t=0 0
m=audio 40926 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv


2020/05/21 11:57:13.362866 192.168.20.15:5060 -> 172.27.160.47:5060
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-6bc0bd877ffd6e88ee83492b0a9c2bb5;received=172.27.160.47
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>;tag=as0f379032
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: IPBX-2.11.0.47(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0



2020/05/21 11:57:13.383867 172.27.160.47:5060 -> 192.168.20.15:5060
ACK sip:51991206@192.168.20.15 SIP/2.0
Call-ID: 761a5e7e19500cec0a745f806374d95c@0:0:0:0:0:0:0:0
Max-Forwards: 70
From: "501" <sip:501@192.168.20.15>;tag=10a6372b
To: <sip:51991206@192.168.20.15>;tag=as0f379032
Via: SIP/2.0/UDP 172.27.160.47:5060;branch=z9hG4bK-393230-6bc0bd877ffd6e88ee83492b0a9c2bb5
CSeq: 2 ACK
Content-Length: 0



2020/05/21 11:56:40.356536 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


2020/05/21 11:56:41.356750 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


2020/05/21 11:56:43.357097 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


2020/05/21 11:56:47.356530 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


2020/05/21 11:56:55.357180 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


2020/05/21 11:57:11.356802 193.168.1.2:5060 -> 190.224.166.135:5060
INVITE sip:51991206@190.224.166.135 SIP/2.0
Via: SIP/2.0/UDP 190.224.166.135:5060;branch=z9hG4bK27bae695
Max-Forwards: 70
From: <sip:541156281700@190.224.166.135>;tag=as42120678
To: <sip:51991206@190.224.166.135>
Contact: <sip:541156281700@190.224.166.135:5060>
Call-ID: 4b7a6bed4afab3e670cb89dc69a97e62@190.224.166.135:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0.47(11.25.3)
Date: Thu, 21 May 2020 14:56:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 1712598779 1712598779 IN IP4 190.224.166.135
s=Asterisk PBX 11.25.3
c=IN IP4 190.224.166.135
t=0 0
m=audio 41146 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
2

What does this mean? The last person who mentioned Issabel was redirected to Issabell support.

There is no 503 response in the log.

The log wasn’t produced by Asterisk, so has to be interpreted from first principles.

There seem to be four IP addresses, so it appears that the log contains entries for which your Asterisk is neither source nor destination.

3

I dont see a final response to the INVITE request

4

Sorry! …I was betrayed by copy paste.

Now i Edit and paste the complete output.

Thanks!

5

You definitely need to provide logs from Asterisk, not some third party capture of the SIP. The reason for the 503 should be in the the Asterisk logging.

6

Based on capture it clearly seems, Telecom is not responding to invite request.
None one here could know and have detail of Telecom/fibertel product information by the way there no public technical information about voice product that they are offering (even on their Pymes and corporations web page info).
Therefore if both doesn’t works together to find out what´s going on, no way to help you.
Moreover if problem could be solve it will be helpful sharing experience information.

7

There are four endpoints and two sessions here. The 503 is associated with one pair of endpoints, and the the timed out INVITE with a different pair.

This is one of the reasons one needs logging from Asterisk, rather than some third party network monitor.

I can see no B side transactions associated with the 503, so the only likely source of information on that will be the VERBOSE and possibly DEBUG logging from Asterisk at the time the decision to send 503 is made.

8

Hi David

Well, I don´t see 4 endpoints I just seem Extension 501(sip:501@192.168.20.15), Asterisk PBX(SIP/2.0/UDP 172.27.160.47) and SiP trunk provider (190.224.166.135)
51991206 is a PSTN number.
So I assumed the following scenario;
501 --→ Asterisk-→ Sip Trunk (Telecom Argentina) < PSTN Network > other carrier -->51991206

Extension 501 → (dial 51991206) to PBX (172.27.160.47) redirect to → Telecom IP (trough SIP Trunk )

So, when extension try to call to 51991206, PBX as IP Trunk it seem no response to invite it send back 503 message to endpoint 501

So at this instances it is very hard to try to debug from Asterisk when causes of the problems becomes between user and the provider (PSTN network), with a many of possible reason as could be; wrong IP (190….) or Telcom have not enabled this user to proceed to make and receive calls and so on with others.

9 
172.27.160.47:5060 -> 192.168.20.15:5060

193.168.1.2:5060 -> 190.224.166.135:5060

Looks like four to me!

10

Thanks, yes you are right.

11

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