# S extension: Inbound call handling **URL:** \ **Category:** Asterisk Support\ **Created:** [September 16, 2011, 7:23pm UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705 "2011-09-16T19:23:06Z")\ **Posts on this page:** 6\ **Page:** 1
**Author:** ![gatorback](https://avatars.discourse-cdn.com/v4/letter/g/74df32/32.png) [@gatorback](https://community.asterisk.org/u/gatorback)\ **Post date:** [September 16, 2011, 7:23pm UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/1 "2011-09-16T19:23:06Z")
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. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [September 16, 2011, 8:50pm UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/2 "2011-09-16T20:50:09Z")
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. ---
**Author:** ![ambiorixg12](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ambiorixg12/32/1887_2.png) [@ambiorixg12](https://community.asterisk.org/u/ambiorixg12)\ **Post date:** [September 17, 2011, 7:18am UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/3 "2011-09-17T07:18:31Z")
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… ---
**Author:** ![gatorback](https://avatars.discourse-cdn.com/v4/letter/g/74df32/32.png) [@gatorback](https://community.asterisk.org/u/gatorback)\ **Post date:** [September 22, 2011, 12:32am UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/4 "2011-09-22T00:32:52Z")
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” \;tag=493e745c442bd52co0 t: sip:1777XXXYYYY@callcentric.com i: [6c6a881c-17e8d46c@192.168.8.110](mailto: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](mailto: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” \;tag=493e745c442bd52co0 To: sip:1777XXXYYYY@callcentric.com;tag=as5f065b8c Call-ID: [6c6a881c-17e8d46c@192.168.8.110](mailto: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” \;tag=493e745c442bd52co0 t: sip:1777XXXYYYY@callcentric.com;tag=as5f065b8c i: [6c6a881c-17e8d46c@192.168.8.110](mailto: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] ---
**Author:** ![dejanst](https://avatars.discourse-cdn.com/v4/letter/d/9fc348/32.png) [@dejanst](https://community.asterisk.org/u/dejanst)\ **Post date:** [September 22, 2011, 5:28pm UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/5 "2011-09-22T17:28:53Z")
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. ---
**Author:** ![gatorback](https://avatars.discourse-cdn.com/v4/letter/g/74df32/32.png) [@gatorback](https://community.asterisk.org/u/gatorback)\ **Post date:** [September 22, 2011, 6:37pm UTC](https://community.asterisk.org/t/s-extension-inbound-call-handling/36705/6 "2011-09-22T18:37:21Z")
@dejanst: Thanks for looking at this question. In the original posting, changing the dialplan statement to: ```auto ``` 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](http://www.callcentric.com/support/device/asterisk/1_6) dtmfmode = rfc2833 context=from-callcentric srvlookup=yes register =\> [1777XXXYYYY:myPassword1@callcentric.com](mailto: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](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](http://host=callcentric.com) defaultuser=1777XXXYYYY ;account=XXXXXXXX, pw=YYYYYYH secret=myPassword1 fromuser=1777XXXYYYY [fromdomain=callcentric.com](http://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](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]](http://supermario-world.blogspot.com/2010/11/asterisk-18-and-native-google-voice.html%5B/code%5D)