[Freeswitch-users] Help! FS Unable to Handle Incoming Calls Through SIP Gateway

Yasuro yasuro at yasuro.com
Sat Nov 13 21:47:48 PST 2010


Michael and other FreeSWITCH gurus:

As per your instruction, I have created logs. Please take a look 
<http://pastebin.freeswitch.org/14479>. It includes FS's messages while 
it initializes itself. The VoIP adapter (SIP gateway, which also works 
as a router)'s IP address is *192.168.11.250* on the LAN side. That of 
the PC which is running FS is *192.168.11.11*. Its firewall function was 
turned off during this testing. The summary of packet exchange between 
the two during this same period is here 
<http://pastebin.freeswitch.org/14480>. Please see my original posting 
(which is included at the end of this message) for the setup of my home LAN.

~250 is sending RTCP receiver report before ~11 starts RTP 
communication. I am wondering if the former is giving up on the call 
because of time out; and, if it is true, if there is any way to have it 
wait longer.

I am completely stuck about this issue and I'll welcome any input you have.

Thanks!

Yasuro

Michael Collins wrote (11/12/2010 4:43 AM):
> Run your same test on FreeSWITCH but turn on sip debugging at the fs_cli:
> sofia global siptrace on
>
> Then make the call and capture the output and put into new pastebin. 
> Hopefully you'll see why the channel is already hungup when it goes to 
> play music.
> -MC
>
> 2010/11/10 Yasuro <yasuro at yasuro.com <mailto:yasuro at yasuro.com>>
>
>     Hi, FreeSWITCH gurus! I need your help!
>
>     First off, I am new to FS and I am new to Internet telephony as
>     well. Heck, I am new to the concept of NAT, UPnP, etc., so please
>     bear with my ignorance.
>
>     I subscribe to a VoIP service at home, with which I get one DID.
>     They supply me a VoIP adapter. Their expected usage is for you to
>     plug in analog phones to the analog phone jacks in the VoIP
>     adapter. However, It also has four Ethernet LAN ports and it acts
>     as a router. You can also access it from the LAN side and register
>     with its built-in SIP gateway.
>
>     What I would like to do is to run FS (Windows version) on one of
>     the Windows PCs, have it register with the SIP gateway, and have
>     it act as an AA or IVR. For testing, I am having it just play music.
>
>     When I tried the same idea with AsterikWin32, it worked just as I
>     had hoped; it answered incoming calls automatically. However, I
>     somehow cannot make it work with FS. I simulate incoming calls to
>     my DID number with Skype's Sypeout. It fails after a short while
>     with such error messages as "network error." It appears the call
>     was never answered.
>
>     FS is assigned an extension number 7 at the gateway. When I call
>     extension 7 from a different extension (at the gateway level, not
>     an extension inside FS), FS does answer the call and I hear music.
>     FS fails to answer only incoming calls from outside.
>
>     I think my FS configuration is fairly standard. I created an
>     external SIP profile for the gateway under
>     conf/sip_profiles/external/ and modified
>     conf/dialplan/public/00_inbound_did.xml so incoming calls to the
>     gateway will be transferred to an extension within FS.
>
>     FS's messages and logs, plus the result of packet captures
>     indicate that FS /thinks /it has answered the call, and goes on to
>     initiate media communication. I see RTP packets going from FS to
>     the SIP gateway. What's different from AsteriskWin32's case is
>     that there are no RTP packets coming back from the SIP gateway to
>     FS. Turning of the firewall of the PC does not seem to change the
>     result in any way.
>
>     For your perusal, I have created the following logs of
>     communication between FS/AsteriskWin32 and the SIP gateway:
>
>         * AsteriskWin32's case
>               o Summary: http://pastebin.freeswitch.org/14457
>               o Details: http://pastebin.freeswitch.org/14460
>         * FreeSWITCh's case
>               o Summary: http://pastebin.freeswitch.org/14462
>               o Details: http://pastebin.freeswitch.org/14463
>               o Log: http://pastebin.freeswitch.org/14465
>
>     The IP address of the SIP gateway is *192.168.11.250*, and that of
>     the PC FS/AsteriskWin32 resides in is *192.168.11.11*. My DID
>     number is masked as ABCDEFGHIJ. I do not know if it gives you any
>     useful information, but those files include the registration
>     phase. /FS's log was taken at a different time/, so it does not
>     entirely match the packet captures.
>
>     I also have the corresponding Pcap files. Please let me know if
>     you need them.
>
>     I am not entirely sure, but I think as far as what I'd like to do
>     is concerned, NAT is not going to be an issue, because the
>     FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP address)
>     are on the same subnet (192.168.11/24). At this time, I do not
>     need to access FS from the Internet.
>
>     Finally, I will give you more details about my setup, which may or
>     may not be relevant to this issue.
>
>     My home LAN is set up this way:
>     http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup.jpg
>     Please note that there are /two layers/ of NAT, and that in the
>     inner layer, two NAT devices exist. I know it looks convoluted,
>     but there are logical reasons for this setup.
>
>     The VoIP service provider only supports the PCMU codec. The music
>     file I prepared for this testing is encoded in PCMU, so codecs
>     will not be an issue.
>
>     Please do not hesitate to ask if you have any questions. Thanks
>     for your help in advance!
>
>     Yasuro
>
>
>
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