S extension: Inbound call handling


#1

I am forced to use this in my dialplan:

[from-callcentric]
exten => 1777XXXXXXX,1,Dial(SIP/101,10)

The ‘s’ extension does not work in this context:
[from-callcentric]
exten => s,1,Dial(SIP/101,10)

The ‘s’ extension does work in this context:
[google-in]
exten => s, 1, Dial(SIP/101, 180, D(:1))

Did I misunderstand how to apply the S-extension? Thanks.


#2

Use the actual number for true Direct in DIaling.

Use s when the service doesn’t tell you the dialled number.

Single number DID (which is called DID in Asterisk circles, but isn’t really) will also need the actual number matching.

Basically, if the invite says:

INVITE

use s

if it says

INVITE @

match the possible numbers.


#3

it is strange i use gtalk and callcentric and the s extensions work fine , but let see the output of your CLI when you use the extension in an incomming call context…


#4

I wonder if this problem is specific to my combination of hardware and software version.

For your viewing pleasure: I turned sip debugging on:

[code]<— SIP read from UDP:204.11.192.31:5060 —>
INVITE sip:1777XXXYYYY@98.82.XX.YY:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.31:5060;branch=z9hG4bK-9eaf37fa80828701369b813b8afbcbf2
f: “firstname lastname” <sip:my number@callcentric.com>;tag=493e745c442bd52co0
t: sip:1777XXXYYYY@callcentric.com
i: 6c6a881c-17e8d46c@192.168.8.110
CSeq: 102 INVITE
Max-Forwards: 13
m: sip:bf4f0eae213a4a04da25854f5fe8e6e5@204.11.192.31:5060;transport=udp
Supported: x-sipura, replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Expires: 240
c: application/sdp
l: 488

v=0
o=17772022880 1 1 IN IP4 204.11.192.31
s=-
c=IN IP4 204.11.192.31
t=0 0
m=audio 55374 RTP/AVP 18 0 2 4 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
— (13 headers 22 lines) —
Sending to 204.11.192.31:5060 (no NAT)
Using INVITE request as basis request - 6c6a881c-17e8d46c@192.168.8.110
No matching peer for ‘my number’ from '204.11.192.31:5060’
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G726-40 for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 204.11.192.31:55374
Looking for 1777XXXYYYY in from-callcentric (domain 98.82.XX.YY:5060)

<— Reliably Transmitting (no NAT) to 204.11.192.31:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.31:5060;branch=z9hG4bK-9eaf37fa80828701369b813b8afbcbf2;received=204.11.192.31
From: “firstname lastname” <sip:my number@callcentric.com>;tag=493e745c442bd52co0
To: sip:1777XXXYYYY@callcentric.com;tag=as5f065b8c
Call-ID: 6c6a881c-17e8d46c@192.168.8.110
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0

<------------>
[Sep 21 20:17:28] NOTICE[3919]: chan_sip.c:21581 handle_request_invite: Call from ‘’ to extension ‘1777XXXYYYY’ rejected because extension not found in context ‘from-callcentric’.
Scheduling destruction of SIP dialog ‘6c6a881c-17e8d46c@192.168.8.110’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:204.11.192.31:5060 —>
ACK sip:1777XXXYYYY@98.82.XX.YY:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.31:5060;branch=z9hG4bK-9eaf37fa80828701369b813b8afbcbf2
f: “firstname lastname” <sip:my number@callcentric.com>;tag=493e745c442bd52co0
t: sip:1777XXXYYYY@callcentric.com;tag=as5f065b8c
i: 6c6a881c-17e8d46c@192.168.8.110
CSeq: 102 ACK
Max-Forwards: 15
l: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘6c6a881c-17e8d46c@192.168.8.110’ Method: ACK
Really destroying SIP dialog ‘7f6d8f1d1f40dc3178696d582fb731df@[c0a8:801:789b:a97f::]’ Method: REGISTER[/code]


#5

It looks like Asterisk does not find extension 1777XXXYYYY in the context.

Please also publish the content of sip.conf and extensions.conf. We need more information.


#6

@dejanst: Thanks for looking at this question. In the original posting, changing the dialplan statement to:

fixed the problem. I have been using the S extension to handle my inbound Google Voice / Jabber calls. Not sure why it works with GV and not with a SIP provider.

Here is an excerpt from my sip.conf:

[code][general]
alwaysauthreject=yes
; 9/6/11
; [ABC] http://www.callcentric.com/support/device/asterisk/1_6
dtmfmode = rfc2833
context=from-callcentric
srvlookup=yes
register => 1777XXXYYYY:myPassword1@callcentric.com/1777XXXYYYY
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas
; http://www.callcentric.com/support/device/asterisk/1_6

[callcentric]
; 9/9/11: attempt to restrict codec
type=peer
context=from-callcentric ;how incoming calls are handle: defined in: extensions.conf
host=callcentric.com
defaultuser=1777XXXYYYY ;account=XXXXXXXX, pw=YYYYYYH
secret=myPassword1
fromuser=1777XXXYYYY
fromdomain=callcentric.com
insecure=invite ;9/6/11
disallow=all
allow=g729

[101]
; [ABC] http://supermario-world.blogspot.com/2010/11/asterisk-18-and-native-google-voice.html
type=friend
host=dynamic
nat=yes
qualify=yes
; context=to-callcentric TEST ONLY, uncomment next line
context=mario-default ; found in exentsions.conf
defaultuser=101
secret=myPassword1
callerid=“SPA2102_L2” <101>
mailbox=101
; [ABC] http://supermario-world.blogspot.com/2010/11/asterisk-18-and-native-google-voice.html[/code]