Hello All<br>
<br>
Thanks for all of your comments. We have done some more work on
this and have noticed the following. We have a pots dial peer for
outbound calls and this works fine. <br>
<br>
Instead of using a pots dial peer we thought that the access-code line
under call manager fallback could be usedinstead. However, we
cannot see any difference when applied. Our syntax is;<br>
<br>
call-manager-fallback<br>
access-code pri 9.T direct-inward-dial<br><br>
Has anyone else seen this? We are using SRST 3.2<br>
<br>
Regards<br>
<br>
Gary<br>
<br>
<br>
<br>
<div><span class="gmail_quote">On 2/23/06, <b class="gmail_sendername">Jonathan Charles</b> <<a href="mailto:jonvoip@gmail.com">jonvoip@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Actually, the DNs that register with SRST are the lines in the first
two positions on the phone... So, if you have bogus numbers there (i.e.
not DIDs), then the phone will register with those numbers.<br><br>You may need to move the DNs around on the phone.
<div><span class="e" id="q_109979cba139394c_1"><br><br><div><span class="gmail_quote">On 2/23/06, <b class="gmail_sendername">Erick Bergquist</b> <<a href="mailto:erickbe@yahoo.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
erickbe@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Also,<br><br>You may need to use translation-rules, connection plar<br>on FXOs, alias under fallback, etc depending on your<br>dial plan. Especially if under normal conditions if<br>your calls route to virtual numbers (not directly to
<br>IP Phone extension that exist in fallback). When in<br>fallback, the IP phones that go in fallback will have<br>their extensions automatically added to router during<br>the time of fallback for up to the max-dn settings
<br>configured, etc per phone model. Any other numbers<br>will need to be handled by the means mentioned above<br>in H.323 to get those calls to route to a phone in<br>fallback.<br><br>You may also want to add 'application mgcpapp' or
<br>'service mgcpapp' to all dial-peers with a analog pots<br>port under them. This is so when in MGCP-mode a<br>inbound call doesn't accidentally terminate on a<br>non-MGCP dial peer and cause problems (dead air, etc).<br>
I have seen this happen. Also make sure you have the<br>ccm-manager fallback-mgcp and global 'call application<br>alternate default' or new global application section<br>with default application specified as alternate.<br>
Without this when MGCP fails the voice-ports on router<br>will not go in fallback and thus won't be available<br>for use by IP phones in fallback.<br><br>--- <a href="mailto:patrick.diener@ita.hsr.ch" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
patrick.diener@ita.hsr.ch
</a> wrote:<br><br>> > Hi<br>> Hi Gary<br>><br>> If the connection to the CallManager is up and<br>> running all call routing<br>> decisions are made by CCM for an MGCP-Gateway. If<br>> the connection to the
<br>> CCM is lost the GW has to rely on its self to make<br>> call routing decisions,<br>> hence you have to configure dial-peers (cause that's<br>> the way call routing<br>> is done on ios platforms)<br>>
<br>> If no dial-peers are configured or an appropriate<br>> dial-peer can not be<br>> found for an incoming call the GW tries to use the<br>> default dial-peer to<br>> create the inbound call-leg. For outbound call-legs
<br>> the default dial-peer<br>> can not be used so a dial-peer MUST be configured.<br>><br>> For more information on ios call routing and<br>> dial-peers:<br>><br><a href="http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080aec.html</a><br>><br>> Patrick<br>><br>> ><br>> > Can anyone clarify the need for dial peers with an
<br>> MGCP gateway running<br>> > SRST<br>> > 3.3. I cannot get the router to dial in or our<br>> without a POTS dial peer<br>> > even though I am using a *<br>> > *dialplan-pattern and a transfer-pattern. Is it
<br>> right that a POTS dial<br>> > peer<br>> > is required? I seem to recall that the router can<br>> donfigure dial peers<br>> > itself on the fly.<br>> ><br>> > Regards<br>> >
<br>> > Gary<br>> > *<br>> > *<br>> > _______________________________________________<br>> > cisco-voip mailing list<br>> > <a href="mailto:cisco-voip@puck.nether.net" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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