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<p>hi Ryan,</p>
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<p><br>
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<p>i have long weekend to resolve the issue. i can't define the hq public ip for global bind because ip sec vpn at spoke use the hq public ip to connect as well. so i have to define dial peer bind command for media and control. will start troubleshooting by
binding the interface at dialpeer level.</p>
<p><br>
</p>
<p>Regards</p>
<p><br>
</p>
<p>Naresh</p>
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<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> Ryan Huff <ryanhuff@outlook.com><br>
<b>Sent:</b> Friday, September 30, 2016 6:01 PM<br>
<b>To:</b> naresh rathore; cisco voip<br>
<b>Subject:</b> Re: one way voice issue</font>
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<p>One-way audio is almost always, always; one side of a call leg's media stream being blocked (ACL, NAT, Firewall ... etc).
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<p><br>
</p>
If the PSTN hears the IP phone but the IP phone cannot hear the PSTN in the northbound direction, then the media stream is simply not getting back to the IP phone. The call sets up fine because signaling travels a different path than media.<br>
<br>
If you're in an 'emergency' type of situation, you can probably work around this for now by checking 'MTP Required' on the CCM SIP trunk and resetting the CCM SIP trunk. This is not ideal and comes with its own set of demons (CCM cpu utilization .. etc); and
if you process a high call volume, this may not even be an option for you. If you attempted this, you'd want to make sure the MRGL the SIP trunk and phones use only has access to media resources that all phone segments can reach and the media resources can
reach all phone segments. This could potentially allow you to 'weather the storm' and get you to a maintenance window where you could tackle it properly.<br>
<br>
Thanks,<br>
<br>
Ryan<br>
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<div id="divRplyFwdMsg" dir="ltr"><font color="#000000" face="Calibri, sans-serif" style="font-size:11pt"><b>From:</b> naresh rathore <nareh84@hotmail.com><br>
<b>Sent:</b> Friday, September 30, 2016 8:41 AM<br>
<b>To:</b> Ryan Huff<br>
<b>Subject:</b> Re: one way voice issue</font>
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<p>hi Ryan</p>
<p><br>
</p>
<p>I havent manually bind the interface, but if cube initiate the session with ITSP, it will use public ip and for CUCM it will use the voice vlan ip.</p>
<p><br>
</p>
<p>Regards</p>
<p><br>
</p>
<p>Naresh</p>
<div style="color:rgb(0,0,0)">
<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="divRplyFwdMsg" dir="ltr"><font color="#000000" face="Calibri, sans-serif" style="font-size:11pt"><b>From:</b> Ryan Huff <ryanhuff@outlook.com><br>
<b>Sent:</b> Friday, September 30, 2016 5:36 PM<br>
<b>To:</b> naresh rathore<br>
<b>Subject:</b> Re: one way voice issue</font>
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<p>Naresh,</p>
<p><br>
</p>
<p>I took a brief look here ... I'm not seeing where you are binding SIP in the HQ cube? That is the first issue I'd tackle. If CCM and the ITSP can see a common interface on the CUBE (i.e MPLS ... etc) then you could do a global bind on that interface under
<i>voice service voip > sip.</i></p>
<p><i><br>
</i></p>
However, if the ITSP trunk really is over the Internet, individual dial peer binding will be more helpful (carrier peers bound to the Internet facing interface, CCM peers bound to the CCM facing interface). Just need to watch that your inbound peers don't overlap
with dial peer binding.<br>
<br>
Thanks,<br>
<br>
Ryan<br>
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<div id="divRplyFwdMsg" dir="ltr"><font color="#000000" face="Calibri, sans-serif" style="font-size:11pt"><b>From:</b> cisco-voip <cisco-voip-bounces@puck.nether.net> on behalf of naresh rathore <nareh84@hotmail.com><br>
<b>Sent:</b> Friday, September 30, 2016 8:21 AM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] one way voice issue</font>
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<p>hi,</p>
<p><br>
</p>
<p>Setup: there are two sites (HQ and spoke). Each site have their own internet and connected to each other via ipsec vpn. CUCM and CUC is at HQ site. this is centralized deployment. the internet gateway at HQ site is also acting as a CUBE and there is sip
trunk up and running between CUBE and ITSP via internet both HQ and spoke site use the HQ cube to call out (using the same ITSP). now site to site calls are working ok. calls from HQ Phones to ITSP and vice versa (mobile/national etc) are working ok. incoming
calls transferred by HQ phone to spoke site phone also works well. the issue is outgoing call from spoke site, when they make outgoing call, the customer can hear their voice but the spoke site phone user cant hear them. any suggestions??. i have attached
the config of both routers.</p>
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</p>
<p>Regards</p>
<p><br>
</p>
<p>Naresh</p>
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