When I tried with FS Server and my receivig system as same system with one IP address.<div>Call Requesting System from other system with second IP address.</div><div>It worked as expected, I can listen the IVR menu.</div><div>
<br></div><div>But the logs says </div><div><div>2011-06-20 15:27:53.054612 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/w</div><div><a href="http://aliaankit1@sip.actionvoip.com:5060">aliaankit1@sip.actionvoip.com:5060</a>] 192.168.1.2 port 24484 -> 194.120.0.106 port</div>
<div> 11772 codec: 0 ms: 20</div></div><div><br></div><div>My Requesting system IP address is correct but my receiving system IP address is 192.168.1.3.</div><div><br></div><div>Is it something related to NAT? Is it something which I dont know about Networking? Is this a random behavior? </div>
<div><br></div><div>Thanks</div><div>Ankit</div><div><br><div class="gmail_quote">On Mon, Jun 20, 2011 at 3:15 PM, ankIT WALiA <span dir="ltr"><<a href="mailto:ankitwalia4u@gmail.com">ankitwalia4u@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hello Guru, MC, Steve and friends,<div><br></div><div>I have pasted my logs on <a href="http://pastebin.freeswitch.org/16555" target="_blank">http://pastebin.freeswitch.org/16555</a></div>
<div><br></div><div>I observed the logs and saw the below line:</div>
<div>2011-06-20 14:57:27.282184 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/external/<a href="http://waliaankit1@sip.actionvoip.com:5060" target="_blank">waliaankit1@sip.actionvoip.com:5060</a>] 192.168.1.2 port 23520 -> 77.72.168.99 port 11994 codec: 8 ms: 20</div>
<div><br></div><div>I think it is telling me that the request is from IP address 192.168.1.2 port 23520 to 77.72.168.99 port 11994.Though, I am using NAT at my broadband modem but my public IP address is also not 77.72.168.99. </div>
<div><br></div><div>I think as told by Steve, the response is going to wrong place.</div><div><br></div><div>Guru, </div><div>I also saw the SIP ALG setting, SIP is enabled.</div><div><br></div><div>Please suggest what I can do in this case. I can try wireshark or pcap, but isnt it evident from the above log that the response is going to the wrong place.</div>
<div><br></div><div>Can it be a problem that my FS server, my call request system and my receiving system is one system only having one network device with IP address 192.168.1.2 ?</div><div><br></div><div>Thanks,</div><div>
Ankit</div><font color="#888888"><div><br></div></font><div><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Mon, Jun 20, 2011 at 3:06 PM, ankIT WALiA <span dir="ltr"><<a href="mailto:ankitwalia4u@gmail.com" target="_blank">ankitwalia4u@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello Guru, MC and friends,<div><br></div><div>Sorry for the delayed reply, I did some changes in my broadband modem, reset the conf <div><div></div><div><br><br><div class="gmail_quote">On Sat, Jun 18, 2011 at 1:58 AM, guru singh <span dir="ltr"><<a href="mailto:grsingh750@gmail.com" target="_blank">grsingh750@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Ankit,<br>
<br>
You could also see if there are SIP ALG settings on the router.<br>
<br>
If by broadband router you mean the DSL modem that providers give,<br>
then you're in a sticky situation. From my experience they are very<br>
unreliable, something that works would break without any apparent<br>
reason or even worse at times things would only work intermittently.<br>
So you just end up racking your brains trying to reproduce and<br>
diagnose, what I'd call "a false problem". Replacing that would reduce<br>
some of your troubles.<br>
<br>
Regards,<br>
<font color="#888888">guru<br>
</font><div><div></div><div><br>
On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br>
> I would get a pcap of both the signaling and the RTP and analyze w/<br>
> wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of<br>
> course, you might start with a console debug log and siptrace on<br>
> <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a> so that we can have a looksee at what is happening<br>
> on the call.<br>
> -MC<br>
><br>
> On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <<a href="mailto:ankitwalia4u@gmail.com" target="_blank">ankitwalia4u@gmail.com</a>> wrote:<br>
>><br>
>> Yes, I am under NAT through my Broadband router.<br>
>> But, if NAT is the issue, then is it possible for both sides able to<br>
>> listen each other voices when I am transferring my call to another ext.<br>
>> which is not IVR<br>
>> Could someone please suggest what I can do to work with NAT or under<br>
>> firewall?<br>
>> Secondly, do you know any other reason for this issue if its not NAT?<br>
>> Thanks<br>
>> Ankit<br>
>><br>
>> On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>> wrote:<br>
>>><br>
>>> Is there any NAT involved? Usually lack of audio means that a NAT router<br>
>>> or firewall is screwing up the RTP media which is sent in a separate UDP<br>
>>> stream. Either it's not being forwarded, or it's being told to go to the<br>
>>> wrong place.<br>
>>><br>
>>> -Steve<br>
>>><br>
>>><br>
>>> On 17 June 2011 13:13, ankIT WALiA <<a href="mailto:ankitwalia4u@gmail.com" target="_blank">ankitwalia4u@gmail.com</a>> wrote:<br>
>>>><br>
>>>> Dear all,<br>
>>>> I have configured an gateway for external profile to receive and make<br>
>>>> calls.<br>
>>>> I have added a file in public directory.<br>
>>>> <include><br>
>>>> <extension name="iptel-inbound"><br>
>>>> <condition field="destination_number"<br>
>>>> expression="^(MY_IPTEL_USERNAME)$"><br>
>>>> <action application="set" data="domain_name=$${domain}"/><br>
>>>> <action application="transfer" data="ExtNo XML default"/><br>
>>>> </condition><br>
>>>> </extension><br>
>>>> </include><br>
>>>> Now, while I am calling to my voip account, I am transferring the call.<br>
>>>> It is getting transferred to the extension also.<br>
>>>> But, If I use normal ext number, I am able to talk both ways. But when I<br>
>>>> use default ext number 5000 for IVR menu. I could not listen to IVR menu.<br>
>>>> Though, I can see in my logs that the audio file are being played one by<br>
>>>> one, but I cant listen.<br>
>>>> Do I need some extra configuration to listen to IVR?<br>
>>>> There may be a gap in my knowledge. Please enlighten.<br>
>>>> Thanks<br>
>>>> Ankit<br>
>>>> --<br>
>>>> Life is like a rose its upto u feel it as its fragrance or thorns<br>
>>>><br>
>>>> _______________________________________________<br>
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>>><br>
>><br>
>><br>
>><br>
>> --<br>
>> Life is like a rose its upto u feel it as its fragrance or thorns<br>
>><br>
>> _______________________________________________<br>
>> Join us at ClueCon 2011, Aug 9-11, Chicago<br>
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</div></div></blockquote></div><br><br clear="all"><br></div></div>-- <br><div>Life is like a rose its upto u feel it as its fragrance or thorns<br>
</div></div>
</blockquote></div><br><br clear="all"><br></div></div>-- <br><div class="im">Life is like a rose its upto u feel it as its fragrance or thorns<br>
</div></div>
</blockquote></div><br><br clear="all"><br>-- <br>Life is like a rose its upto u feel it as its fragrance or thorns<br>
</div>