[Freeswitch-users] Incoming call from registered gateway with DID in To: not URI

[Steven Ayre]

Sofia debugging shows this if it helps identify the problem at all:

nta: INVITE (6915472) going to existing leg
nta: timer shortened to 200 ms
nua: nua_stack_process_request: entering
soa_init_offer_answer(static::0x2346220) called
soa_set_remote_sdp(static::0x2346220, (nil), 0x233a90e, 521) called
nua(0x22f3260): INVITE server: error parsing SDP
nua: nua_invite_server_respond: entering
soa_clear_remote_sdp(static::0x2346220) called
tport_tsend(0x23720e0) tpn = UDP/80.93.165.111:5060
tport_resolve addrinfo = 80.93.165.111:5060
tport_by_addrinfo(0x23720e0): not found by name UDP/80.93.165.111:5060
tport_vsend(0x23720e0): 572 bytes of 572 to udp/80.93.165.111:5060
tport_vsend returned 572
nta: sent 400 Bad Session Description for INVITE (6915472)

-Steve


On 8 January 2011 16:11, Steven Ayre <steveayre at gmail.com> wrote:
> From the source code it looks like that means the SDP is invalid - do
> you see the problem in those packet traces?
>
> Warm Regards,
> -Steve
>
>
> On 8 January 2011 16:08, Steven Ayre <steveayre at gmail.com> wrote:
>>> One of their guys said by email they tested on Git within the last few
>>> weeks and found calls were hanging up after 2m40s due to something
>>> about how we handle timers... I'm going to test that next. Were there
>>> any versions in Git recently that might have had problems?
>>
>> There does appear to be an issue here.
>>
>> Calls of 4mins duration with no problems with enable-timer=false
>>
>> With enable-timer=true the call times out. The reason is they send an
>> INVITE and we reply 400 Bad Session Description.
>> http://pastebin.freeswitch.org/14957 shows those two packets.
>>
>> -Steve
>>
>>
>>
>> On 8 January 2011 15:56, Steven Ayre <steveayre at gmail.com> wrote:
>>> <param name="extension" value="auto_to_user"/> was what did the trick
>>> though. Thanks! :)
>>>
>>> There's a minor issue from it where the trunk shows unregistered on
>>> their website (they expect the Contact header to be username at ip") but
>>> incoming calls are still sent through fine so it doesn't actually
>>> matter at all. They can still see FS registered on their switch.
>>>
>>> I've added the configuration to the SIP Provider Examples section of the Wiki.
>>>
>>> One of their guys said by email they tested on Git within the last few
>>> weeks and found calls were hanging up after 2m40s due to something
>>> about how we handle timers... I'm going to test that next. Were there
>>> any versions in Git recently that might have had problems?
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 8 January 2011 02:50, Brian West <brian at freeswitch.org> wrote:
>>>> Also if you set the extension to auto_to_user it'll do it also.
>>>>
>>>> /b
>>>>
>>>> On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
>>>>
>>>>> I'm currently handling it with:
>>>>>
>>>>>  <param name="context" value="numbergroup"/>
>>>>>
>>>>> and:
>>>>>
>>>>>  <context name="numbergroup">
>>>>>    <extension name="numbergroup">
>>>>>      <condition field="destination_number" expression="^USERNAME$">
>>>>>        <action application="transfer" data="${sip_to_user} XML default"/>
>>>>>      </condition>
>>>>>    </extension>
>>>>>  </context>
>>>>>
>>>>> Is there a better way of handling it? Such as a param that tells Sofia
>>>>> to use the To not the URI to populate destination_number?
>>>>>
>>>>> -Steve
>>>>
>>>>
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>>>>
>>>
>>
>