# Solved -- Incoming calls from Sipgate **URL:** \ **Category:** Asterisk Support\ **Created:** [September 15, 2009, 5:04am UTC](https://community.asterisk.org/t/solved-incoming-calls-from-sipgate/29008 "2009-09-15T05:04:14Z")\ **Posts on this page:** 1\ **Page:** 1
**Author:** ![Gokee2](https://avatars.discourse-cdn.com/v4/letter/g/c0e974/32.png) [@Gokee2](https://community.asterisk.org/u/Gokee2)\ **Post date:** [September 15, 2009, 5:04am UTC](https://community.asterisk.org/t/solved-incoming-calls-from-sipgate/29008/1 "2009-09-15T05:04:14Z")
Hello everyone, ``` I am running asterisk (1.4.21.2~dfsg-3) on Debian stable. I have SFLphone on my workstation to make calls through. I have a ipcop gateway that the asterisk server and the workstation are behind. I got a sipgate account to make calls with. So far I have got SLFphone connected to asterisk and it can make outgoing calls. But I can't get incoming calls. Sipgate always shows my phone as offline and test calls do not show up in sip debug or on my gateway. I configured my NAT setup according to [forums.digium.com/viewtopic.php?t=7854](http://forums.digium.com/viewtopic.php?t=7854) ``` My ipcop configuration looks like: ```auto UDP DEFAULT IP :5060 - 5082 => 192.168.0.47 : 5060 - 5082 Asterisk --- SIP signaling UDP DEFAULT IP :10000 - 20000 => 192.168.0.47 :10000 - 20000 Asterisk --- RTP traffic TCP DEFAULT IP :5060 - 5082 => 192.168.0.47 : 5060 - 5082 Asterisk --- SIP signaling ``` I have a file called sip\_nat.conf ```auto NAT=Yes externip=24.113.159.168 localnet=192.168.0.0/255.255.255.0 ``` In my sip.conf I have: ```auto [general] NAT=Yes externip=24.113.159.168 localnet=192.168.0.0/255.255.255.0 [1000] type=friend context=phones host=dynamic secret=its_passtastic register => my-sip-id:my-sip-password@sipgate/my-sip-id [sipgate] type=peer nat=yes canreinvite=no ; nat=1 ; externip=24.113.159.168 secret=my-sip-password insecure=invite username=my-sip-id defaultuser=my-sip-id fromuser=my-sip-id context=sipgate_in ;context=incoming_calls fromdomain=sipgate.com host=sipgate.com outboundproxy=proxy.live.sipgate.com qualify=yes disallow=all allow=ulaw dtmfmode=rfc2833 ;#include sip_nat.conf ``` In my extensions.conf I have: ```auto [globals] [general] static=yes ; NAT=Yes ; externip=24.113.159.168 ; externip= ***.***. ***.*** ; localnet=192.168.0.0/255.255.255.0 [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [incoming_calls] exten => _X.,1.NoOp() exten => _X.,n,Dial(SIP/1000) ;sexten => my-sip-id,1,Dial(SIP/1000) [outgoing_calls] ;exten => _X.,1,NoOp() ;exten => _X.,n,Dial(SIP/my_service_provider/${EXTEN}) exten => _X.,1,Set(CALLERID(num)=SIP-ID) exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg) [internal] exten => 500,1,Verbose(1|Echo test application) exten => 500,n,Echo() exten => 500,n,Hangup() exten => 1000,1,Verbose(1|Extension 1000) exten => 1000,n,Dial(SIP/1000,30) exten => 1000,n,Hangup() [phones] include => internal include => outgoing_calls ; Stuff from sipgate [sipgate_in] exten => my-sip-id,1,Dial(SIP/1000); <-- instead of extension you should define the corresponding peer exten => my-sip-id,n,Hangup ; [sipgate_out] ; exten => _X.,1,Set(CALLERID(num)=SIP-ID) ; exten => _X.,2,Dial(SIP/${EXTEN}@sipgate,30,trg) ; exten => _X.,3,Hangup ``` Whenever asterisk starts it gives messages like: ```auto [Sep 14 18:05:46] NOTICE[29237] cdr.c: CDR simple logging enabled. [Sep 14 18:05:46] NOTICE[29237] loader.c: 160 modules will be loaded. [Sep 14 18:05:46] WARNING[29237] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Sep 14 18:05:46] NOTICE[29237] res_odbc.c: Adding ENV var: INFORMIXSERVER=my_special_database [Sep 14 18:05:46] NOTICE[29237] res_odbc.c: Adding ENV var: INFORMIXDIR=/opt/informix [Sep 14 18:05:46] NOTICE[29237] res_odbc.c: res_odbc loaded. [Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: Starting AEL load process. [Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Sep 14 18:05:46] NOTICE[29237] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load [Sep 14 18:05:46] WARNING[29237] chan_iax2.c: Unable to open IAX timing interface: No such file or directory [Sep 14 18:05:46] ERROR[29237] chan_vpb.cc: No Voicetronix cards detected [Sep 14 18:05:46] NOTICE[29248] chan_sip.c: Peer 'sipgate' is now Reachable. (78ms / 2000ms) ``` With sip debugging on I get messages in the console like: ```auto <--- SIP read from 204.155.28.10:5060 ---> SIP/2.0 200 OK Record-Route: Via: SIP/2.0/UDP 24.113.159.168:5060;branch=z9hG4bK4eb124d1;rport=5060 From: "asterisk" ;tag=as41100537 To: ;tag=ddf9a9f0de4595ec12441b9d8d3ee250.00b0 Call-ID: 2197a5e855066ec040ff95e04c51897e@24.113.159.168 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '2197a5e855066ec040ff95e04c51897e@24.113.159.168' Method: OPTIONS Reliably Transmitting (NAT) to 204.155.28.10:5060: OPTIONS sip:sipgate.com SIP/2.0 Via: SIP/2.0/UDP 24.113.159.168:5060;branch=z9hG4bK70c65f61;rport From: "asterisk" ;tag=as5c8af6a9 To: Contact: Call-ID: 787c25a77625508a4de5fbbe08a721f3@24.113.159.168 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 15 Sep 2009 04:42:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ``` Edit: Fixed. My register command was not in the general section.