[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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