Hello Guru, MC and friends, Sorry for the delayed reply, I did some changes in my broadband modem, reset the conf On Sat, Jun 18, 2011 at 1:58 AM, guru singh <grsingh750@gmail.com> wrote:
Hi Ankit,
You could also see if there are SIP ALG settings on the router.
If by broadband router you mean the DSL modem that providers give, then you're in a sticky situation. From my experience they are very unreliable, something that works would break without any apparent reason or even worse at times things would only work intermittently. So you just end up racking your brains trying to reproduce and diagnose, what I'd call "a false problem". Replacing that would reduce some of your troubles.
Regards, guru
On Sat, Jun 18, 2011 at 12:35 AM, Michael Collins <msc@freeswitch.org> wrote:
I would get a pcap of both the signaling and the RTP and analyze w/ wireshark. Find out if the RTP is actually leaving FreeSWITCH or not. Of course, you might start with a console debug log and siptrace on pastebin.freeswitch.org so that we can have a looksee at what is happening on the call. -MC
On Fri, Jun 17, 2011 at 9:07 AM, ankIT WALiA <ankitwalia4u@gmail.com> wrote:
Yes, I am under NAT through my Broadband router. But, if NAT is the issue, then is it possible for both sides able to listen each other voices when I am transferring my call to another ext. which is not IVR Could someone please suggest what I can do to work with NAT or under firewall? Secondly, do you know any other reason for this issue if its not NAT? Thanks Ankit
On Fri, Jun 17, 2011 at 7:05 PM, Steven Ayre <steveayre@gmail.com>
wrote:
Is there any NAT involved? Usually lack of audio means that a NAT
router
or firewall is screwing up the RTP media which is sent in a separate UDP stream. Either it's not being forwarded, or it's being told to go to the wrong place.
-Steve
On 17 June 2011 13:13, ankIT WALiA <ankitwalia4u@gmail.com> wrote:
Dear all, I have configured an gateway for external profile to receive and make calls. I have added a file in public directory. <include> <extension name="iptel-inbound"> <condition field="destination_number" expression="^(MY_IPTEL_USERNAME)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="ExtNo XML default"/> </condition> </extension> </include> Now, while I am calling to my voip account, I am transferring the
call.
It is getting transferred to the extension also. But, If I use normal ext number, I am able to talk both ways. But when I use default ext number 5000 for IVR menu. I could not listen to IVR menu. Though, I can see in my logs that the audio file are being played one by one, but I cant listen. Do I need some extra configuration to listen to IVR? There may be a gap in my knowledge. Please enlighten. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns
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