<div dir="ltr">Please post one of your sip.conf phone configs, so we can have a look.<br><div class="gmail_extra"><br clear="all"></div><div class="gmail_extra">Alyed<br><br><br></div><div class="gmail_extra"><div class="gmail_quote">
2013/8/2 Carlos Chavez <span dir="ltr"><<a href="mailto:cursor@telecomabmex.com" target="_blank">cursor@telecomabmex.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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On 8/1/13 9:17 PM, Michael L. Young wrote:<br>
> ----- Original Message -----<br>
>> From: "Carlos Chavez" <<a href="mailto:cursor@telecomabmex.com" target="_blank">cursor@telecomabmex.com</a>> To:<br>
>> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a> Sent: Thursday, August 1, 2013<br>
>> 8:41:19 PM Subject: [asterisk-users] External sip phones register<br>
>> with the servers IP...<br>
>><br>
>> We have just updated our office server to Asterisk 11.4.0 from<br>
>> 1.8.15 and internally everything is working fine. The problem we<br>
>> are having is that we cannot use any external phone connected<br>
>> through the Internet. This used to work fine with 1.8 but since<br>
>> the upgrade whenever you register any phone from an outside<br>
>> network the phone tries to register using the servers internal<br>
>> IP.<br>
>><br>
>> I endo up having something like this:<br>
>><br>
>> Sending to <a href="http://187.163.93.235:58545" target="_blank">187.163.93.235:58545</a> (no NAT) -- Registered SIP '2003'<br>
>> at <a href="http://192.168.2.50:58545" target="_blank">192.168.2.50:58545</a> Reliably Transmitting (no NAT) to<br>
>> <a href="http://192.168.2.50:58545" target="_blank">192.168.2.50:58545</a>: OPTIONS sip:2003@192.168.2.50:58545;ob<br>
>> SIP/2.0 Via: SIP/2.0/UDP<br>
>> 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From:<br>
>> "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.2.50" target="_blank">sip:asterisk@192.168.2.50</a>>;tag=as4ed13172 To:<br>
>> <sip:2003@192.168.2.50:58545;ob> Contact:<br>
>> <<a href="http://sip:asterisk@192.168.2.50:5060" target="_blank">sip:asterisk@192.168.2.50:5060</a>> Call-ID:<br>
>> <a href="http://46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060" target="_blank">46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060</a> CSeq: 102<br>
>> OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013<br>
>> 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,<br>
>> SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer<br>
>> Content-Length: 0<br>
>><br>
>> I really cannot understand what is wrong, I have checked my<br>
>> sip.conf configuration and it is the same as in past versions.<br>
>> externaddr and localnet are set to the proper values. Any<br>
>> ideas?<br>
><br>
> Did you look at the CHANGES file? There are new settings for NAT.<br>
> If you are using the same settings as in 1.8, there is a posiblity<br>
> that you will have problems depending on what settings you have<br>
> (which you did not include in this message).<br>
><br>
> Also, I would recommend 11.5 since there was a one-way audio issue<br>
> fixed related to using the two new NAT settings.<br>
><br>
I have tried with all nat variations and I get the same result. I<br>
upgraded to 11.5 yesterday but same problem. External phones still<br>
register using the servers internal IP. IAX works fine.<br>
<br>
- --<br>
Telecomunicaciones Abiertas de México S.A. de C.V.<br>
Carlos Chávez Prats<br>
Director de Tecnología<br>
+52-55-91169161 ext 2001<br>
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</blockquote></div><br></div></div>