if you have an option to disable sip Uri dialling it might help. it helps with the polygons. On May 18, 2011 10:02 AM, "Laurent Schweizer" < [email protected]> wrote: > Hi, > > > > the outbound proxy is already set > > > > and If I check the INVITE trace (both bellow) of the snom , both call are going to the same IP 192.168.20.35:5060 > > > > > > > > OK > > Sent to tcp:192.168.20.35:5060 at 18/5/2011 08:40:26:611 (1211 bytes): > > INVITE sip:[email protected];user=phone SIP/2.0 > Via: SIP/2.0/TCP 192.168.20.123:5060;branch=z9hG4bK-yghazja4btk6;rport > From: "Florent Schreiber" <sip:[email protected]>;tag=zg08oi15tw > To: <sip:[email protected];user=phone> > Call-ID: 062bc43c547d-jwvgv0ifyvv8 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: <sip:[email protected]:5060 ;transport=tcp;line=114zk55p>;reg-id=1 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.18 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 393 > > > > > > > > NOT OK > > Sent to tcp:192.168.20.35:5060 at 18/5/2011 08:43:00:410 (1199 bytes): > > INVITE sip:[email protected];user=phone SIP/2.0 > Via: SIP/2.0/TCP 192.168.20.123:5060;branch=z9hG4bK-vcrc9trmbg2x;rport > From: "Florent Schreiber" <sip:[email protected]>;tag=ep3aoo43pn > To: <sip:[email protected];user=phone> > Call-ID: a02bc43c8150-pw2zlxd5365e > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: <sip:[email protected]:5060 ;transport=tcp;line=114zk55p>;reg-id=1 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom821/8.4.18 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 395 > > > > De : Joegen Baclor [mailto:[email protected]] > Envoyé : mercredi 18 mai 2011 12:50 > À : Discussion list for users of sipXecs software > Objet : Re: [sipx-users] calls directly send to the domain of the call > > > > @paul > > I don't think an alias would do it. His problem is the call "Dial last caller" functionality does not send the call to sipx but to the b2bua directly which is not routable from his phone. > > @laurent > > Most phones have a configurable parameter to "always use outbound proxy". Look for the equivalent setting in snom and set the IP of sipx as the outbound proxy. That should force all calls coming out of snom to traverse sipx. > > > > On 05/18/2011 06:09 PM, [email protected] wrote: > > Hi Laurent, > > I am not sure whether this will work, but it's worth a try: > > In the SipX GUI under <System> go to <Domain>, then click <Add Alias> and add 95.128.80.93 as alias. > > Paul > > > > > From: > > "Laurent Schweizer" <mailto:[email protected]> < [email protected]> > > > To: > > "'Discussion list for users of sipXecs software'" <mailto: [email protected]> <[email protected]> > > > Date: > > 18-05-2011 10:31 > > > Subject: > > Re: [sipx-users] calls directly send to the domain of the call > > > Sent by: > > [email protected] > > > > _____ > > > > > no no I don't what to do URI dialing. > > for me if all call can go like the first one it's ok . > the problem is when I receive an incoming call, the snom phone is storing the domain of the From header and not only the phone number . (and teh domain is the not the sipX domain ) > > so when I call back this number he place the call with the domain and the second scenario happens. > > so if you have any solution to configure the snom phone to not store or not use the received domain or in sipX to change the domain during the incoming call it will solve my problem. > > Laurent > > De : Todd Hodgen [ <mailto:[email protected]> mailto: [email protected]] > Envoyé : mercredi 18 mai 2011 09:53 > À : 'Discussion list for users of sipXecs software' > Objet : Re: [sipx-users] calls directly send to the domain of the call > > Can you please explain what it is you are trying to do? It seems your call goes out when dialed using conventional methods. Are you trying to use some sort of URI dialing and bypass the use of trunks with the second call. If you are, there are methods for doing that, but it requires some different configurations. Frankly, I’m not sure what issue you are discovering that needs resolution. > > If you look at the session for your first call, it is routing itself through the local proxy, the local bridge, and then to your caller. This call is much different than trying to dial directly to an IP address. I would recommend looking at the wiki at wiki.sipfoundry.org for directions on setting up internet or URL dialing. Unless I’m missing something here. > > From: [email protected] [ <mailto: [email protected]> mailto: [email protected]] On Behalf Of Laurent Schweizer > Sent: Wednesday, May 18, 2011 12:07 AM > To: 'Discussion list for users of sipXecs software' > Subject: Re: [sipx-users] calls directly send to the domain of the call > > Hello all, > > I have done the following test , from the same phone I have placed 2 calls: > the first to a simple number 0215522001 and the snom phone as added the domain > the second to a number with a domain [email protected] > > and this is the path of the first call, as you can see all is ok he go to the correct path > > > > > now for the second call, the invite is directly send to the IP 95.128.80.93 > > > > > > regarding the registration of the snom phone, (done with the auto configuration of sipX ) I have: > > > > > so if he must check the dial plan to route the call, why for the second he don't do it ? > > regards > > Laurent > > De : Todd Hodgen [ <mailto:[email protected]> mailto: [email protected]] > Envoyé : mercredi 18 mai 2011 01:39 > À : 'Discussion list for users of sipXecs software' > Objet : Re: [sipx-users] calls directly send to the domain of the call > > I think you are confusing the principles here. > > Your outgoing call will be placed on a trunk based on the dial plan that you set up. The phone is registered with the Proxy, and based on permissions it has, it can dial out on multiple dial plans in the system. It will grab the first route that it has permissions enabled for, and has a dial pattern match. You will have complete control over that path in your configuration. > > Register the phone with the Domain name, that is critical to successful deployment. Once registered, and receiving calls, move to your dial plans and ensure you can get calls out the way that you want. > > From: [email protected] [ <mailto: [email protected]> mailto: [email protected]] On Behalf Of Laurent Schweizer > Sent: Tuesday, May 17, 2011 4:07 PM > To: 'Discussion list for users of sipXecs software' > Subject: Re: [sipx-users] calls directly send to the domain of the call > > > I understand that sipX work like a proxy so he must forward the call to the > destination present in the domain part. > > As phone I have snom phone configured with the default auto-provisioning and > they seems to store the domain present in the >From header and not only the > number > > so the question is, if I receive a call from a trunk and the phone store not > only the number but also the domain, when I call back this call he will > place the new call with the domain so he will use the same trunk as > incoming but maybe I need to use another trunk for outgoing calls. > > is that possible to configure the snom to only store the number part and not > the domain ? > > Laurent > > > > -----Message d'origine----- > De : Worley, Dale R (Dale) [ <mailto:[email protected]> mailto: [email protected]] > Envoyé : mardi 17 mai 2011 22:08 > À : Discussion list for users of sipXecs software > Objet : Re: [sipx-users] calls directly send to the domain of the call > > ________________________________________ > From: [email protected] > [[email protected]] On Behalf Of Laurent Schweizer > [[email protected]] > > if I start the call with a correct domain (the Sipx domain) all is ok. > > how can I solve this issue ? > ________________________________________ > > Ensure that the call is sent with the correct domain. > > sipX follows the SIP standards. If the call's "request URI" is > <sip:[email protected]> "sip:[email protected]", then sipX is directed > to send the call to host 1.2.3.4, and host 1.2.3.4 is responsible for > interpreting "foo". > > Either the phone is mis-configured so that it is sending an INVITE with > incorrect domain in the request URI, > or there is some NAT/firewall device that thinks it should modify SIP > INVITEs that pass through it. > > Dale > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _____ > > No virus found in this message. > Checked by AVG - <http://www.avg.com/> www.avg.com > Version: 10.0.1375 / Virus Database: 1509/3643 - Release Date: 05/17/11 > > > > _____ > > No virus found in this message. > Checked by AVG - <http://www.avg.com/> www.avg.com > Version: 10.0.1375 / Virus Database: 1509/3644 - Release Date: 05/17/11_______________________________________________ > sipx-users mailing list > [email protected] > List Archive: <http://list.sipfoundry.org/archive/sipx-users/> http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > >
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