Congestion tone

Asterisk Asterisk Support
1

I have this context in my extensions.conf:

[custom-DigiVoice-in] ;;prepare for callback and DISA exten => 6042885566,1,NoOp(Incoming call from DigitalVoice #6042885566) exten => 6042885566,2,Congestion exten => 6042885566,3,Hangup
When calling 6042885566 it should give the caller Congestion tone but it does not work. It just keep ringing! The CLI shows:

[quote] – Executing [6042885566@mydomain-incoming:1] NoOp(“SIP/6042885566-00137e68”, “Incoming call from DigitalVoice #6042885566”) in new stack
– Executing [6042885566@mydomain-incoming:2] Congestion(“SIP/6042885566-00137e68”, “”) in new stack
== Spawn extension (mydomain-incoming, 6042885566, 2) exited non-zero on ‘SIP/6042885566-00137e68’
[/quote]

Did I miss something here?

Please help!

2

Hi,

just to have the correct syntax,I think it should be Congestion() and Hangup()

3

thats because you never accept the call. Your provider sends the call to your * box, which replies that it’s busy. Not sure why the provider keeps ringing though, perhaps something else is registered to your account?

either way, put Answer() between NoOP() and Congestion(). That will accept the call and answer it, and you will hear busy tone. Note that that busy tone is just audio from *, not an actual busy signal, so as far as the provider is concerned it’s a good established answered call.

4

Thanks both of you for the replies!

Last night it was too late and I didn’t give more detail… sorry! :blush:

I made changes as you recommend but still it won’t give me the congestion tone.

Here is more detail about my setup:

I am running a slimmed Asterisk 1.4 on a Linksys NSLU2 with Unslug installed. Here is the package link:
ipkgfind.nslu2-linux.org/details … l=&format=

I am trying to setup callback and give the caller DISA. I have done this before on another * box by following the instruction at here, .

So the first thing I needed is give the caller a congestion tone, so the caller won’t be charged for the success connection.

Maybe there is something that is required to run Congestion() successfully. I just don’t know. I saw the CLI shows that the application executed with no error. I tried to load all the modules in to * and I got the same result.

Any idea?

Thanks.

5

I aslo tried Playtones(congestion), not working! :cry:

voip-info.org/wiki/index.php … Congestion

6

well if you want to avoid establishing a call then you are doing it the right way, just Congestion() and nothing else. However it sounds like possibly your provider is not taking this to heart and is continuing to let the other end ring… post a sip debug?

7

Here is the SIP debug from CLI:

8

try busy() instead of congestion() and post a sip debug?

also just for kicks post the relevant sip.conf sections…

9

use busy() instead of congestion() …

[quote]NSLU2*CLI>
<— SIP read from 209.17.160.129:5060 —>
INVITE sip:6042885566@24.87.119.52 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK1812bbda;rport
From: “16042079158” sip:16042079158@209.17.160.129;tag=as58c1d6d6
To: sip:6042885566@24.87.119.52
Contact: sip:16042079158@209.17.160.129
Call-ID: 7e467b9018856df22fa1c7205046dfdd@209.17.160.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 02 Jan 2007 01:19:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 12757 12757 IN IP4 209.17.160.129
s=session
c=IN IP4 209.17.160.129
t=0 0
m=audio 14516 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
— (12 headers 13 lines) —
Sending to 209.17.160.129 : 5060 (NAT)
Using INVITE request as basis request - 7e467b9018856df22fa1c7205046dfdd@209.17.160.129
Found peer 'DigiVoice5561’
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 209.17.160.129:14516
Found description format G729 for ID 18
Found description format GSM for ID 3
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.17.160.129:14516
Looking for 6042885566 in mydomain-incoming (domain 24.87.119.52)
list_route: hop: sip:16042079158@209.17.160.129

<— Transmitting (NAT) to 209.17.160.129:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK1812bbda;received=209.17.160.129;rport=5060
From: “16042079158” sip:16042079158@209.17.160.129;tag=as58c1d6d6
To: sip:6042885566@24.87.119.52
Call-ID: 7e467b9018856df22fa1c7205046dfdd@209.17.160.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:6042885566@24.87.119.52
Content-Length: 0

<------------>
– Executing [6042885566@mydomain-incoming:1] NoOp(“SIP/6042885566-0013e1f8”, “Incoming call from DigitalVoice #6042885566”) in new stack
– Executing [6042885566@mydomain-incoming:2] Busy(“SIP/6042885566-0013e1f8”, “”) in new stack

<— Transmitting (NAT) to 209.17.160.129:5060 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK1812bbda;received=209.17.160.129;rport=5060
From: “16042079158” sip:16042079158@209.17.160.129;tag=as58c1d6d6
To: sip:6042885566@24.87.119.52;tag=as57e212a4
Call-ID: 7e467b9018856df22fa1c7205046dfdd@209.17.160.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
== Spawn extension (mydomain-incoming, 6042885566, 2) exited non-zero on 'SIP/6042885566-0013e1f8’
Really destroying SIP dialog ‘7e467b9018856df22fa1c7205046dfdd@209.17.160.129’ Method: INVITE
NSLU2*CLI>
<— SIP read from 209.17.160.129:5060 —>
ACK sip:6042885566@24.87.119.52 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK1812bbda;rport
From: “16042079158” sip:16042079158@209.17.160.129;tag=as58c1d6d6
To: sip:6042885566@24.87.119.52;tag=as57e212a4
Contact: sip:16042079158@209.17.160.129
Call-ID: 7e467b9018856df22fa1c7205046dfdd@209.17.160.129
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
— (9 headers 0 lines) —
NSLU2*CLI>
<— SIP read from 209.17.160.129:5060 —>
OPTIONS sip:6042885566@24.87.119.52 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK4b647335
From: “asterisk” sip:asterisk@209.17.160.129;tag=as03bc0f35
To: sip:6042885566@24.87.119.52
Contact: sip:asterisk@209.17.160.129
Call-ID: 704eb31c295fc3477b3305ec38b1d701@209.17.160.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 02 Jan 2007 01:19:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Looking for 6042885566 in mydomain-incoming (domain 24.87.119.52)

<— Transmitting (NAT) to 209.17.160.129:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK4b647335;received=209.17.160.129
From: “asterisk” sip:asterisk@209.17.160.129;tag=as03bc0f35
To: sip:6042885566@24.87.119.52;tag=as2531eb47
Call-ID: 704eb31c295fc3477b3305ec38b1d701@209.17.160.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:24.87.119.52
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘704eb31c295fc3477b3305ec38b1d701@209.17.160.129’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.1.28:5061:
OPTIONS sip:dynaguy@192.168.1.28:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK522e3a29;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as1e55b2a0
To: sip:dynaguy@192.168.1.28:5061
Contact: sip:asterisk@192.168.1.77
Call-ID: 1e1947153e600a5d2dc12c5257f688a3@192.168.1.77
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 02 Jan 2007 01:17:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

NSLU2*CLI>
<— SIP read from 192.168.1.28:5061 —>
SIP/2.0 200 OK
To: sip:dynaguy@192.168.1.28:5061;tag=e61a960c8b46b485i1
From: “asterisk” sip:asterisk@192.168.1.77;tag=as1e55b2a0
Call-ID: 1e1947153e600a5d2dc12c5257f688a3@192.168.1.77
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK522e3a29
Server: Linksys/PAP2-3.1.12(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘1e1947153e600a5d2dc12c5257f688a3@192.168.1.77’ Method: OPTIONS
[/quote]

10

sip.conf:

11

I found the problem! It is not the Asterisk’s problem.

When I test it I use a telephone which registered on another VSP through a PAP2. It seems this VSP I am using have some problem.

I just tried with my cellphone (stupid me, should thought about this earlier. :blush: ), voipstunt, skype., they all get the congestion tone!

What is the real reason behind this?

Anyway, at least I know that my Asterisk is working fine.

Thank you, IronHelix.

And, HAPPY NEW YEAR TO YOU! :wink:

12

Hi,

Its good that you found a solution. Bot sure if this is related, but here goes:

With SIP users, when they wanted to dial, I needed them to enter a PIN first, without answering the call, so they wont be charged. So I used Background() with the NOANSWER
option, and it ONLY worked if i did a Progress() first, then it worked, so try that. Also, no matter what I did, it worked on ALL my SIP phones except eyebean exten.
So, there you go :smile: