Gafachi doesn't make a gateway do they? The best way to do this, and I have done this but it's been a long time, is to use two physical gateways:
Sipx<->gateway (audiocodes, patton, fxo models only),<->physical phone cord connected to your voip provider provided device, which is an FXS gateway and connects to whatever network(s) you need to get it's dial tone, since the RJ11 phone cord won't care. We used to do this when we started to avoid LD charges on the phone system and it wasn't really hard to do. We since graduated to SIP trunks with Ingate after trying a few other things and have been very happy. The caveat with SIP trunks is getting the right provider. We had to cycle through two of them, because they are not all equal. Or so as Michael said and use OpnSBC or Ingate. If you are trying to go into a production mode I don't think 3.11.8 is stated to be nearly stable yet, there's a lot being done in the code. I tried OpenSBC a long time ago and went with Ingate for siptrunks. I think there is the beginning of a wiki page started for OpenSBC: http://sipx-wiki.calivia.com/index.php/SipXecs-OpenSBC The issue you have is NAT traversal. You are trying to define a siptrunk in your dial plan byy the providers IP address. IF you have a public IP address on your phones and your sip system with no firewalls, this would actually work. Because you are on a private network though, you can't traverse NAT natively with sipx, until the next major relase ships. So you need to: 1. Use an ATA to connect to a provider (Vonage, whatever) and connect the phone line to a FXO gateway that works with sipx 2. Use a SBC or SIP aware firewall to connect to the sip trunk, and define that device (not the provider directly) and use it as a gateway within sipx. 3. Use a standard POTS line with an FXO gateway. SIP doesn't traverse NAT through a firewall easily. Just forwarding port 5060 won't help. The media uses high ports and is random, plus a picture is worth a thousand words. http://freshmeat.net/articles/view/2079/ Hope this helps. Tony >>> Chad Leigh - Pengar Enterprises Inc <[EMAIL PROTECTED]> 11/27/08 2:42 AM >>> Ok I installed sipXecs. I finally got a few different models of aastra phones set up on some extensions and can dial between them. Please send me to the RFM if that is the answer. I have been searching the wiki and Google for a couple of hours. My sipXecs server is on an internal network but is also defined as being in the DMZ so is directly accessible from outside. For the few hours I played with FreeSWITCH that worked fine and I actually got calls to go in and out from my zaptel based phone (SIP phones were in a UPS truck somewhere at the time). For outgoing: I defined a SIP Trunk as a gateway (Gafachi) under Gateways. I turned on a few of the default dial plan rules (long distance, toll free, international, local) and for local defined my local area code as a prefix to be prepended. The gateway is set in these rules. I dial my call phone or my vonage phone from one of the extensions and get a call failure. Looking in SIPVIEWER I get this See the end of this message for SIP details INVITE 100 Trying ACK 302 Moved Temporarily over and over It does not seem to be trying the gateway. -- For incoming I do not see anywhere to define incoming plans. I would like to be able to specify which incoming calls (different toll free ) ring which extensions etc. For now, if I call the Gafachi hosted toll free number from my cell or vonage phones, this is what is happening here with the SIPVIEWER SIP Details for outgoing: ------ INVITE Time: 2008-11-27T07:11:34.436832Z Frame: 1 sipregistrar.xml:293 Source: 192.168.99.174:5060 Dest: denwa.pengar.com-SipRegistrar INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: <sip:192.168.99.174:5060;lr> Via: SIP/2.0/TCP 192.168.99.174;branch=z9hG4bK- sipXecs-00ed7a4d7d1596c6ab6f5762e1d338e42c00 Via: SIP/2.0/UDP 192.168.99.211;branch=z9hG4bK18f4666d9 Max-Forwards: 20 Content-Length: 283 To: 8015272405 <sip:[EMAIL PROTECTED]> From: sip:[EMAIL PROTECTED];tag=52ea57199c88327 Call-Id: [EMAIL PROTECTED] Cseq: 208943983 INVITE Supported: timer Allow-Events: talk,hold,conference Allow: NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: <sip:[EMAIL PROTECTED];transport=udp> Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 Date: Thu, 27 Nov 2008 07:11:34 GMT X-Sipx-Spiral: true Route: <sip:192.168.99.174:5070;transport=tcp;lr> v=0 o=MxSIP 0 1943314278 IN IP4 192.168.99.211 s=SIP Call c=IN IP4 192.168.99.211 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=silenceSupp:on - - - - ------ 100 Trying Time: 2008-11-27T07:11:34.437676Z Frame: 2 sipregistrar.xml:294 Source: denwa.pengar.com-SipRegistrar Dest: 192.168.99.174:5060 SIP/2.0 100 Trying From: sip:[EMAIL PROTECTED];tag=52ea57199c88327 To: 8015272405 <sip:[EMAIL PROTECTED]> Call-Id: [EMAIL PROTECTED] Cseq: 208943983 INVITE Via: SIP/2.0/TCP 192.168.99.174;branch=z9hG4bK- sipXecs-00ed7a4d7d1596c6ab6f5762e1d338e42c00 Via: SIP/2.0/UDP 192.168.99.211;branch=z9hG4bK18f4666d9 Content-Length: 0 ------ ACK Time: 2008-11-27T07:11:34.442689Z Frame: 3 sipregistrar.xml:296 Source: 192.168.99.174:5060 Dest: denwa.pengar.com-SipRegistrar ACK sip:[EMAIL PROTECTED] SIP/2.0 Route: <sip:192.168.99.174:5070;transport=tcp;lr> Contact: <sip:[EMAIL PROTECTED];transport=udp> From: sip:[EMAIL PROTECTED];tag=52ea57199c88327 To: 8015272405 <sip:[EMAIL PROTECTED]> Call-Id: [EMAIL PROTECTED] Cseq: 208943983 ACK Max-Forwards: 20 Via: SIP/2.0/TCP 192.168.99.174;branch=z9hG4bK- sipXecs-00ed7a4d7d1596c6ab6f5762e1d338e42c00 Content-Length: 0 ------ 302 Moved Temporarily Time: 2008-11-27T07:11:34.447287Z Frame: 4 sipregistrar.xml:297 Source: denwa.pengar.com-SipRegistrar Dest: 192.168.99.174:5060 SIP/2.0 302 Moved Temporarily Contact: <sip:[EMAIL PROTECTED] > From: sip:[EMAIL PROTECTED];tag=52ea57199c88327 To: 8015272405 <sip:[EMAIL PROTECTED]> Call-Id: [EMAIL PROTECTED] Cseq: 208943983 INVITE Via: SIP/2.0/TCP 192.168.99.174;branch=z9hG4bK- sipXecs-00ed7a4d7d1596c6ab6f5762e1d338e42c00 Via: SIP/2.0/UDP 192.168.99.211;branch=z9hG4bK18f4666d9 User-Agent: sipXecs/3.10.2 sipXecs/registry (Linux) Date: Thu, 27 Nov 2008 07:11:34 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE Accept-Language: en Supported: gruu Content-Length: 0 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
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