# Call routing problem **URL:** \ **Category:** Asterisk Support\ **Created:** [May 18, 2007, 7:09pm UTC](https://community.asterisk.org/t/call-routing-problem/14513 "2007-05-18T19:09:28Z")\ **Posts on this page:** 7\ **Page:** 1
**Author:** ![ironoxid](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ironoxid/32/1044_2.png) [@ironoxid](https://community.asterisk.org/u/ironoxid)\ **Post date:** [May 18, 2007, 7:09pm UTC](https://community.asterisk.org/t/call-routing-problem/14513/1 "2007-05-18T19:09:28Z")
Hi, before all, excuse me for my bad english šŸ˜’ Iā€™m trying to configure asterisk with an italian SIP provider called Skypho. This provider use the same host for incoming and outgoing calls ([voip.eutelia.it](http://voip.eutelia.it)). Iā€™ve configured a peer like this [myprovider-out] username secret host [voip.eutelia.it](http://voip.eutelia.it) nat etc and a register line. SIP SHOW REGISTRY tells that asterisk is connected to receive incoming calls from [voip.eutelia.it](http://voip.eutelia.it) and in fact, it does. outgoing calls work very well. But when I receive a call from outside there is a problemā€¦ incoming call is redirect (i think) another time to my voip provider throught myprovider-out peer ā˜¹ So I canā€™t catch incoming calls by dialplane extensions. Caller receives an occupied tone and nothing else. SIP DEBUG reports: [code]\<ā€” SIP read from 83.211.227.21:5060 ā€”\> INVITE sip:9999@NET.NET.NET.MYIP SIP/2.0 Record-Route: sip:83.211.227.21;ftag=3149D7E4-1DD9;lr=on Record-Route: sip:83.211.227.13;ftag=3149D7E4-1DD9;lr=on Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK71b6.1b070b62.0 Via: SIP/2.0/UDP 62.94.88.138:5060;branch=z9hG4bK15D316115D From: sip:33975\*\*\*\*\*@62.94.88.138;tag=3149D7E4-1DD9 To: sip:04441837XXX@voip.eutelia.it Call-ID: [DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138](mailto:DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138) Supported: rel1xx,timer,replaces CSeq: 102 INVITE Max-Forwards: 8 Remote-Party-ID: sip:33975\*\*\*\*\*@62.94.88.138;party=calling;screen=yes;privacy=off Contact: sip:33975\*\*\*\*\*@62.94.88.138:5060 Expires: 180 Content-Type: application/sdp Content-Length: 448 v=0 o=CiscoSystemsSIP-GW-UserAgent 8937 3170 IN IP4 62.94.88.138 s=SIP Call c=IN IP4 62.94.88.139 t=0 0 m=audio 17686 RTP/AVP 18 8 0 4 3 125 101 19 c=IN IP4 62.94.88.139 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:125 X-CCD/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:19 CN/8000 a=direction:passive \<-------------\> ā€” (17 headers 19 lines) ā€” Sending to 83.211.227.21 : 5060 (NAT) Using INVITE request as basis request - [DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138](mailto:DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138) Found peer ā€˜04441837XXX-outā€™ \<ā€” Reliably Transmitting (NAT) to 83.211.227.21:5060 ā€”\> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK71b6.1b070b62.0 Via: SIP/2.0/UDP 62.94.88.138:5060;branch=z9hG4bK15D316115D From: sip:33975\*\*\*\*\*@62.94.88.138;tag=3149D7E4-1DD9 To: sip:04441837XXX@voip.eutelia.it;tag=as6f10dd32 Call-ID: [DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138](mailto:DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138) CSeq: 102 INVITE User-Agent: Asterisk\_Eut Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=ā€œ[voip.eutelia.it](http://voip.eutelia.it)ā€, nonce="6465c8c6" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ā€˜DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138ā€™ in 32000 ms (Method: INVITE) webdev_CLI\> webdev_CLI\> \<ā€” SIP read from 83.211.227.21:5060 ā€”\> ACK sip:9999@NET.NET.NET.MIOIP SIP/2.0 Max-Forwards: 15 Record-Route: sip:83.211.227.21;ftag=3149D7E4-1DD9;lr=on Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.13;branch=z9hG4bK71b6.1b070b62.0 From: sip:33975\*\*\*\*\*@62.94.88.138;tag=3149D7E4-1DD9 Call-ID: [DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138](mailto:DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138) To: sip:04441837XXX@voip.eutelia.it;tag=as6f10dd32 CSeq: 102 ACK Content-Length: 0[/code] Note: If i change myprovider-out from peer to user incoming calls work perfectly but I canā€™t initiate an outgoing call (user canā€™t be used to connect to my provider). Any help would be appreciated Thank you! Giulio edit: debian testing - asterisk (1:1.2.13~dfsg-2) ---
**Author:** ![mmacvicar](https://avatars.discourse-cdn.com/v4/letter/m/e8c25b/32.png) [@mmacvicar](https://community.asterisk.org/u/mmacvicar)\ **Post date:** [May 18, 2007, 10:08pm UTC](https://community.asterisk.org/t/call-routing-problem/14513/2 "2007-05-18T22:08:59Z")
It sounds like your inbound context is broken. Youā€™ll need to post your contexts from sip.conf for someone to tell if that is the problem. Mark MacVicar ---
**Author:** ![ironoxid](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ironoxid/32/1044_2.png) [@ironoxid](https://community.asterisk.org/u/ironoxid)\ **Post date:** [May 19, 2007, 10:13am UTC](https://community.asterisk.org/t/call-routing-problem/14513/3 "2007-05-19T10:13:16Z")
Broken context itā€™s possible (Iā€™m not an expert, this is my first experience with asterisk šŸ˜„ ) but why when I set peer to user my context/dialplane works perfectly? šŸ˜• I canā€™t post now my .conf files sorry, Iā€™ll post they next monday, but I can try to explain situation for the moment. ```auto [general] context=from-ext ;this is my context for incoming calls realm=voip.eutelia.it [phone1,2,N] context=from-sip ;this is my context for outgoing calls ``` My peer, myprovider-out, itā€™s also on `from-sip` context. Regards Giulio ---
**Author:** ![ironoxid](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ironoxid/32/1044_2.png) [@ironoxid](https://community.asterisk.org/u/ironoxid)\ **Post date:** [May 21, 2007, 7:25am UTC](https://community.asterisk.org/t/call-routing-problem/14513/4 "2007-05-21T07:25:05Z")
sip.conf [code][general] context=from\_ext ; per la gestione delle chiamate entranti da VoIP [realm=voip.eutelia.it](http://realm=voip.eutelia.it) ; Dominio del servizio VoIP SIP port=5060 ; Porta UDP per la segnalazione SIP bindport=5060 bindaddr=0.0.0.0 ; Indirizzo IP di ascolto (generico) srvlookup=yes ; Abilita il DNS SRV per chiamate uscenti defaultexpirey=330 ; timer di registrazione degli account SIP useragent=Asterisk\_Eut nat = yes ; PARAMETRI IP localnet=192.168.5.0/255.255.255.0 ; subnet e mask della LAN externip=82.105.X.X ; indirizzo IP pubblico ; DEFINIZIONE DEI CODEC IN USO disallow=all ;allow=g729 ; da attivare in modalitƃ 'passthroughā€™ allow=alaw allow=ulaw callerid = Unknow ; register =\> ::@voip.eutelia.it/ [register=04441837XXX:MYPASS:04441837XXX@voip.eutelia.it](mailto:register=04441837XXX:MYPASS:04441837XXX@voip.eutelia.it)/04441837XXX [04441837XXX-out] context = from\_sip type=peer secret=77/p1d.i username=04441837XXX fromuser=04441837XXX [host=voip.eutelia.it](http://host=voip.eutelia.it) nat = yes dtmfmode=inband qualify=yes regcontext=from\_sip canrenvite=no [xlite1] type=friend context = from\_sip username=xlite1 callerid="Giulio" host=dynamic nat=no canreinvite=no disallow=all allow=gsm ; GSM consumes far less bandwidth than G711-u (ulaw) allow=ulaw allow=alaw [/code] estensions.conf [code][general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2 ; Trunk interface [from-sip-external] include =\> from\_ext [from\_ext] exten =\> 04441837XXX,1, Answer exten =\> 04441837XXX,n,BackGround(auth-thankyou) exten =\> 04441837XXX,n, Hangup [from\_sip] exten =\> 1010,1, Dial(SIP/xlite1,30,t) exten =\> 1011,1, Dial(SIP/xlite2,30,t) exten =\> 1012,1, Dial(SIP/claudio,30,t) exten =\> \_X.,1,Answer exten =\> \_X.,n,Set(LANGUAGE()=it) exten =\> \_X.,n,Wait(1) exten =\> \_X.,n,BackGround(beep) exten =\> \_X.,n,Dial(SIP/${EXTEN}@04441837XXX-out,30,t) [/code] When I receive a call from outside there is a problemā€¦ incoming call is redirected (i think) another time to my voip provider throught 04441837XXX-out peer If i change myprovider-out from peer to user incoming calls work perfectly but I canā€™t initiate an outgoing call (user canā€™t be used to connect to my provider). Any help would be appreciated Thank you! Giulio ---
**Author:** ![mmacvicar](https://avatars.discourse-cdn.com/v4/letter/m/e8c25b/32.png) [@mmacvicar](https://community.asterisk.org/u/mmacvicar)\ **Post date:** [May 21, 2007, 8:34pm UTC](https://community.asterisk.org/t/call-routing-problem/14513/5 "2007-05-21T20:34:26Z")
I need to get my terminology straight, apologies. I think I see your problem. You need different contexts defined for your inbound and outbound calls, otherwise dialing in will behave as if you are dialing out. Your outbound definition looks good. ## Your inbound definition ā€œxlite1ā€ in sip.conf should refer to a incoming context in extensions.conf and not ā€œfrom\_sipā€. For example: ## [xlite1] type=friend context = incoming ā€¦ ## in extensions.conf, create an ā€œincomingā€ context and have it dial your phone. For example if your Zap channel is 4, a really simple incoming context might be: ## [incoming] exten =\> s,1,Answer ; answer call exten =\> YOURNUMBER,2,Dial(Zap/4,20) exten =\> YOURNUMBER,3,Hangup If you can get that to work then youā€™re on the right track and itā€™s just a matter of getting your dialplan to work the way you want it. This page may explain it better [voip-info.org/wiki/index.php ā€¦ g+sip.conf](http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf) Also, as a note about security make sure you donā€™t past your username and secret in these forums. It looks like you might have missed a few. Mark MacVicar ---
**Author:** ![ironoxid](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ironoxid/32/1044_2.png) [@ironoxid](https://community.asterisk.org/u/ironoxid)\ **Post date:** [May 22, 2007, 10:57pm UTC](https://community.asterisk.org/t/call-routing-problem/14513/6 "2007-05-22T22:57:03Z")
Hmmm youā€™re right eheh, thank you to let me know my -bip- šŸ˜„ Anyway I tried your solution and nothing change ā˜¹ Asterisk seems to completly ignore my dialplanā€¦ in fact it doesnā€™t say ā€œhey boy, i canā€™t find a dialplan!ā€ on sip debugā€¦ itā€™s strangeā€¦ Maybe a problem with destination of call? from sip debug: ```auto ``` Destinationā€™s host and peerā€™s host are the sameā€¦ ([voip.eutelia.it](http://voip.eutelia.it)) I think that asterisk wants to work as a router. So it simply skips my dialplan and rewrite a new request for destinationā€™s host to route call by peerā€¦ from sip debug: ```auto Sending to 83.211.227.21 : 5060 (NAT) <<<----- (aka voip.eutelia.it) Using INVITE request as basis request - DD8E9292-3C211DC-99F9A8A4-2570551E@62.94.88.138 Found peer '04441837XXX-out' ``` Why asterisk uses a peer to route an incoming call? Itā€™s possible? Can I stop this behavior? Thanks a lot Giulio ---
**Author:** ![mmacvicar](https://avatars.discourse-cdn.com/v4/letter/m/e8c25b/32.png) [@mmacvicar](https://community.asterisk.org/u/mmacvicar)\ **Post date:** [May 23, 2007, 4:43pm UTC](https://community.asterisk.org/t/call-routing-problem/14513/7 "2007-05-23T16:43:34Z")
I donā€™t know the answers to your questions. Iā€™d have to see your new sip.conf and extensions.conf to diagnose the issue. there are some useful links here for getting started with asterisk: [asterisk.org/support/get-started](http://www.asterisk.org/support/get-started) Including ā€œAsterisk: The Future of Telephonyā€ from Oā€™Reilly Media, which has a good section on setting up your first dialplan in Chapter 5: [asteriskdocs.org/modules/tin ā€¦ .php?id=11](http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11) Mark MacVicar