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

Peter Steinbach lists at telefaks.de
Sun Nov 14 16:01:07 PST 2010


Thanks David,

missed the "C".

But anyway, I am wondering why .250 sends a BYE right after reception of
"Destination unreachable (Port unreachable)".

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
Theo-Geisel-Strasse 25
D 61250 Usingen, Germany
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



David Ponzone schrieb:
> Peter,
>
> that's not RTP, but RTCP.
> The RTCP port is always RTP+1.
>
> David Ponzone  Direction Technique
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>
> Le 14/11/2010 à 22:31, Peter Steinbach a écrit :
>
>> Hello Yasuro,
>>
>> what I am wondering about is: Freeswitch is offering port 16670 in its
>> SDP message of the SIP OK message, and 192.168.11.250 is trying to send
>> RTP packets to 16671 instead. But this Freeswitch port is not open. So
>> 192.168.11.250 closes the connection: See
>>    217 49.351287   192.168.11.250        5005    192.168.11.11       
>> 16671    RTCP     Receiver Report   Source description   
>>    218 49.351369   192.168.11.11         5005    192.168.11.250       
>> 16671    ICMP     Destination unreachable (Port unreachable)
>>
>>
>>
>> -- 
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> Theo-Geisel-Strasse 25
>> D 61250 Usingen, Germany
>> mailto:lists (att) telefaks.de <http://telefaks.de>
>> Internet: www.telefaks.de <http://www.telefaks.de>
>>
>>
>>
>> Yasuro schrieb:
>>> 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>
>>>> <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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