[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
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service Client IPeva
> tel: 0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pièces jointes sont confidentiels et établis
> à l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autorisée est interdite. Tout message électronique est
> susceptible d'altération. /*/IPeva/*/ décline toute responsabilité au
> titre de ce message s'il a été altéré, déformé ou falsifié. Si vous
> n'êtes pas destinataire de ce message, merci de le détruire
> immédiatement et d'avertir l'expéditeur./
> /
> /
>
>
>
> 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
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
More information about the FreeSWITCH-users
mailing list