<html><head></head><body bgcolor="#FFFFFF"><div>Check for any kind of SIP interference from the end user's router. <br><br>Thanks,<div>--Warren Selby, dCAP</div></div><div><br>On Oct 14, 2011, at 2:38 PM, Adam Robins <<a href="mailto:arobins@PharmaCentra.com">arobins@PharmaCentra.com</a>> wrote:<br><br></div><div><span></span></div><blockquote type="cite"><div>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks I will do that. The user is remote, so I must first RDP into her home network and do it from her PC.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:asterisk-users-bounces@lists.digium.com]
<b>On Behalf Of </b>Danny Nicholas<br>
<b>Sent:</b> Friday, October 14, 2011 3:35 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] Problem with outbound dialing from remote phone<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I use 501’s here and I can pull up the settings by typing
<a href="http://1.2.3.4/index.htm%20-%20where%201.2.3.4">http://1.2.3.4/index.htm - where 1.2.3.4</a> is the IP address of the phone. If you can do that, perhaps something there will be of use to you.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
<a href="mailto:[mailto:asterisk-users-bounces@lists.digium.com]">[mailto:asterisk-users-bounces@lists.digium.com]</a>
<b>On Behalf Of </b>Adam Robins<br>
<b>Sent:</b> Friday, October 14, 2011 2:26 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Problem with outbound dialing from remote phone<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Turned on “sip set debug peer 1234”. I see the qualify messages. I see when she calls me on my internal extension. I see no SIP messages at all when she
calls my cell phone.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I understand what Doug and Eric are saying. I need to get into the phone’s web interface to see how it is programmed just to validate that the phone is still
as I programmed it. What is strange is:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoListParagraph" style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><span style="mso-list:Ignore">a.<span style="font:7.0pt "Times New Roman"">
</span></span></span><!--[endif]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Phone “A” can dial local extensions but not external, so I send her Phone “B”.<o:p></o:p></span></p>
<p class="MsoListParagraph" style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><span style="mso-list:Ignore">b.<span style="font:7.0pt "Times New Roman"">
</span></span></span><!--[endif]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Phone “B” cant dial outbound at all<o:p></o:p></span></p>
<p class="MsoListParagraph" style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><span style="mso-list:Ignore">c.<span style="font:7.0pt "Times New Roman"">
</span></span></span><!--[endif]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Both phones were successfully tested for both call types prior to shipping and were not in any way reconfigured subsequent to testing.<o:p></o:p></span></p>
<p class="MsoListParagraph" style="text-indent:-.25in;mso-list:l0 level1 lfo2"><!--[if !supportLists]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><span style="mso-list:Ignore">d.<span style="font:7.0pt "Times New Roman"">
</span></span></span><!--[endif]--><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I have not modified the digitmap is sip.cfg in years, and even so, entering the number and then pressing ‘Dial’ doesn’t work either.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
<a href="mailto:[mailto:asterisk-users-bounces@lists.digium.com]">[mailto:asterisk-users-bounces@lists.digium.com]</a>
<b>On Behalf Of </b>Sammy Govind<br>
<b>Sent:</b> Friday, October 14, 2011 2:34 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Problem with outbound dialing from remote phone<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<p class="MsoNormal">Hey,<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Can you enable sip trace for that particular sip extension. This sounds weird that while other INVITES from the phone are reaching but the external extensions are filtered. If there are no invites for external calls only then more chances
are that the phone is using some dial pattern(phonebook help) etc like Doug and Eric said. Sometimes in asterisk console I don't see anything in logs if the Sip extensions' context don't contain the number that is being dialled<o:p></o:p></p>
<div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">Do you've access to any phone debugging console?<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><span style="color:white">Sounds like problem is somewhere around "She" :p j/k . </span><o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">--<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">Sammy.<o:p></o:p></p>
<div>
<p class="MsoNormal">On Fri, Oct 14, 2011 at 10:34 PM, Adam Robins <<a href="mailto:arobins@pharmacentra.com">arobins@pharmacentra.com</a>> wrote:<o:p></o:p></p>
<p class="MsoNormal">The phone was originally provisioned from an FTP server when it was inside our network. Once in the field, the phone no longer has access to that server (it could if I wanted it to). It boots using the last known config, which worked
before shipping. I've been doing it this way for 5+ years. This is the first problem of its kind. I can get into the phone by RDPing to the users laptop over VPN and then accessing the phone web interface. I will try that.<br>
<br>
Please remember, I've already tried two phones, both of which worked fine at another remote location prior to shipping, having been programmed from good config files. The first one actually worked fine at this remote location for a period of time and then
suddenly "went bad".<o:p></o:p></p>
<div>
<div>
<p class="MsoNormal"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Eric Wieling<br>
Sent: Friday, October 14, 2011 1:16 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone<br>
<br>
I am assuming you are using a provisioning server.<br>
<br>
If the phone is running firmware 3.2 or earlier you can access the phone web interface and confirm the dialplan active on the phone is the same as what you set in the config file on the server.<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Adam Robins<br>
Sent: Friday, October 14, 2011 12:51 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone<br>
<br>
I've already done that. Both phones worked fine in a different remote location just prior to shipping.<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Doug Lytle<br>
Sent: Friday, October 14, 2011 12:48 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone<br>
<br>
<br>
Adam Robins wrote:<br>
> No change, thanks<br>
<br>
Well,<br>
<br>
In the long run, it may just be easier to send her out a replacement phone and ask for that one back, so you can test in house.<br>
<br>
Doug<br>
<br>
<br>
--<br>
<br>
Ben Franklin quote:<br>
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"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."<br>
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