# Newbie question about Call from '' to extension 'asterisk' r **URL:** \ **Category:** Asterisk Support\ **Created:** [February 20, 2011, 11:18pm UTC](https://community.asterisk.org/t/newbie-question-about-call-from-to-extension-asterisk-r/34228 "2011-02-20T23:18:52Z")\ **Posts on this page:** 4\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [February 20, 2011, 11:18pm UTC](https://community.asterisk.org/t/newbie-question-about-call-from-to-extension-asterisk-r/34228/1 "2011-02-20T23:18:52Z")
Hi group, I am trying to handle my first incoming SIP call through Asterisk and I have already hit a snag. I looked over the Internet for other poor souls such as myself with similar problems and did not succeed in fixing the problem. OBJECTIVE: I want basic call handling with the system answering, playing a prompt and then hanging up. SET-UP: I have the following plan defined in extensions.conf: [default] include =\> incoming [incoming] exten =\> s,1,Wait(1) exten =\> s,n,Answer() exten =\> s,n(goodbye),Playback(vm-goodbye) exten =\> s,n(end),Hangup() In SIP.conf, I have the following: udpbindaddr=0.0.0.0:5090 [asterisk] type=incoming host=dynamic secret=burp context=incoming deny=0.0.0.0/0 When using my SIP phone (Kphone), I get the following output in the terminal (verbose level): [Feb 20 17:34:08] NOTICE[2687]: chan\_sip.c:21355 handle\_request\_invite: Call from ‘’ to extension ‘asterisk’ rejected because extension not found in context ‘incoming’. [Feb 20 17:34:38] WARNING[2687]: chan\_sip.c:3386 retrans\_pkt: Retransmission timeout reached on transmission [1212204217@192.168.1.102](mailto:1212204217@192.168.1.102) for seqno 4259 (Critical Response) – See doc/sip-retransmit.txt. Packet timed out after 31999ms with no response And in the full log (not much more information): [Feb 20 17:34:38] VERBOSE[2687] netsock2.c: == Using SIP RTP CoS mark 5 [Feb 20 17:34:38] NOTICE[2687] chan\_sip.c: Call from ‘’ to extension ‘asterisk’ rejected because extension not found in context ‘incoming’. [Feb 20 17:34:40] WARNING[2687] chan\_sip.c: Retransmission timeout reached on transmission [1027795433@192.168.1.102](mailto:1027795433@192.168.1.102) for seqno 342 (Critical Response) ---
**Author:** ![dejanst](https://avatars.discourse-cdn.com/v4/letter/d/9fc348/32.png) [@dejanst](https://community.asterisk.org/u/dejanst)\ **Post date:** [February 21, 2011, 7:10am UTC](https://community.asterisk.org/t/newbie-question-about-call-from-to-extension-asterisk-r/34228/2 "2011-02-21T07:10:12Z")
Check the settings of the Softphone, your problem is there. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [February 21, 2011, 12:10pm UTC](https://community.asterisk.org/t/newbie-question-about-call-from-to-extension-asterisk-r/34228/3 "2011-02-21T12:10:42Z")
I see nothing wrong with the soft phone. type=incoming is new to me. Which version of Asterisk is this. The device entry seems to have matched correctly, and asterisk is correctly reporting that there is no extension called asterisk in the incoming context. You need to add that extension. ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [April 12, 2011, 4:07am UTC](https://community.asterisk.org/t/newbie-question-about-call-from-to-extension-asterisk-r/34228/4 "2011-04-12T04:07:40Z")
Had the same problem. Solution is to use the vmexten setting in your sip extension config. From the sip.conf sample file `;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defaults to "asterisk ` Your softphone is subscribing for a voicemail notification, and the reply from Asterisk is telling it that the voicemail extension is ‘asterisk’ - the default. When you use your softphone to access voicemail (sipdroid for me), it dials the provided extension.