[Freeswitch-users] No Voice for default IVR ext 5000
ankIT WALiA
ankitwalia4u at gmail.com
Mon Jun 20 13:45:37 MSD 2011
Hello Guru, MC, Steve and friends,
I have pasted my logs on http://pastebin.freeswitch.org/16555
I observed the logs and saw the below line:
2011-06-20 14:57:27.282184 [DEBUG] sofia_glue.c:2594 AUDIO RTP
[sofia/external/waliaankit1 at sip.actionvoip.com:5060] 192.168.1.2 port 23520
-> 77.72.168.99 port 11994 codec: 8 ms: 20
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.
I think as told by Steve, the response is going to wrong place.
Guru,
I also saw the SIP ALG setting, SIP is enabled.
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.
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 ?
Thanks,
Ankit
On Mon, Jun 20, 2011 at 3:06 PM, ankIT WALiA <ankitwalia4u at gmail.com> wrote:
> 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 at 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 at 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 at 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 at 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 at 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
>> >>>>
>> >>>> _______________________________________________
>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>> >>>>
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>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
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>> >>
>> >>
>> >>
>> >> --
>> >> Life is like a rose its upto u feel it as its fragrance or thorns
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
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>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
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>>
>> _______________________________________________
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>> http://www.cluecon.com 877-7-4ACLUE
>>
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>
>
>
> --
> Life is like a rose its upto u feel it as its fragrance or thorns
>
--
Life is like a rose its upto u feel it as its fragrance or thorns
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