I'm actually wondering if using mod_skinny (ie SCCP signalling) may be more appropriate for these Cisco phones. I know that the SIP implementation on the 79XX phones has been criticised and I'm now seeing a new issue (or at least, didn't notice it before when testing after a reboot of the Cisco phone):<div>
<br></div><div>1. 7970G phone (ext. 1001) registers against FreeSWITCH and can dial out and receives bridged inbound calls (and calls from a test softphone on my Android phone (ext. 1000));</div><div>2. After an undetermined period of time (I'm not sure exactly, but it is probably at least 24 hours), calls to 1001 from 1000 receive a generic "<span style="font-family:sans-serif;font-size:13px;line-height:19.046875px;background-color:rgb(255,255,255)">480 Temporarily Unavailable" response and hang up;</span></div>
<div><font face="sans-serif"><span style="line-height:19.046875px">3. Reboot of 1001 resolves issue; restart of FreeSWITCH server does not fix issue.<br></span></font><br>The other option would be to test against a more recent build on a spare linux laptop an see if this issue is fixed in later version?<br>
<br>J<br><br><div class="gmail_quote">On 28 June 2013 15:22, Chris Ferreira <span dir="ltr"><<a href="mailto:chris@gonumina.com" target="_blank">chris@gonumina.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF"><div>I'm not all that familiar with Internode, but I would think you could also have the handset register one of the other "lines" to an extension on Internode that's tied to the VM box. I could see this working on VoIP.ms which I'm familiar with.<br>
<br><div>___________________</div><div>Mobile Reply</div></div><div><br>On Jun 27, 2013, at 8:18 PM, Jeremy Moschner <<a href="mailto:jeremy.moschner@gmail.com" target="_blank">jeremy.moschner@gmail.com</a>> wrote:<br>
<br></div><div>
</div><blockquote type="cite"><div><span style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">Hi Everyone</span><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">
<br></div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">I am running an embedded FreeSWITCH installation (on OpenWRT, opkg version 1.0.7-2) and have a single Cisco 7970G that registers fine and can make and recieve calls (most of the time).</div>
<div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif"><br></div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">
However, one of my SIP providers (Internode) also offers a voicemail system (useful if I have my phone turned off and means I don't have to run a voicemail module on my installation).</div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">
<br></div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">What I would like to find out is if it is possible for the 7970G (or indeed any other endpoint) to query/receive notification from Internode's voicemail system and display a MWI (or some sort of notification on the phone) when a message is there.</div>
<div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif"><br></div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">
Now, from my reading, I believe that this is accomplished through either SIP NOTIFY or SUBSCRIBE messages, but I am still quite confused about how it all works. I am unsure if FreeSWITCH needs to "relay" this information to the endpoint, or if is configurable via the MWI-Account parameter (ie. the directory user registers to receive these updates directly from the SIP server).</div>
<div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif"><br></div><div style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">
I imagine I'll need to provide a bit more detail, so please let me know where to start.</div>
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<br>
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