[Freeswitch-users] Hangup problem/SIP BYE lacking authentication
Anthony Minessale
anthony.minessale at gmail.com
Fri Oct 31 08:16:25 PDT 2008
Yes i mean add it to the dial string inside the {}
it only will work if the channel with the variable set is tied to the FS
session obj.
sofia_reg.c 1122 is where it all happens
so if session is null there the var code won't work.
you can add some debug code there and try to figure out what's wrong.
On Fri, Oct 31, 2008 at 10:06 AM, Wellie Chao <wchao at yahoo.com> wrote:
> I tried the following in conf/dialplan/extensions/7_inbound.xml:
>
> <extension name="broadview_inbound_9325">
> <condition field="destination_number"
> expression="^12675379325|2675379325$">
> <action application="export" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="1001"/>
> </condition>
> </extension>
>
> Also tried the following in conf/dialplan/public.xml:
>
> <extension name="public_did_broadview">
> <condition field="destination_number"
> expression="^(12675379324|2675379324|12675379325|2675379325)$">
> <action application="export" data="sip_use_gateway=broadview"/>
> <action application="transfer" data="$1 XML default"/>
> </condition>
> </extension>
>
> Neither helped. When you say add it to the dial string directly that calls
> it, I'm not sure what you mean (I know the general format of
> {var_name=var_value}, so that's not my question). Do you mean add it in
> front of the 1001 as the target of the transfer?
>
> By the way, hangup DOES work properly if I create another gateway and name
> it 64.115.128.6. However, I'd love to get it working without having to
> create a duplicate gateway with a non-intuitive name. It's definitely a lot
> better than nothing to do it that way, but I'd prefer to have it work with
> the sip_use_gateway scheme you mention. I'm assuming I'm just doing
> something wrong with how sip_use_gateway should be specified in the XML
> configuration files. Can you tell what I am doing wrong?
>
> On Fri, 31 Oct 2008, Anthony Minessale wrote:
>
> Date: Fri, 31 Oct 2008 09:49:18 -0500
>>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Reply-To: freeswitch-users at lists.freeswitch.org
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking
>> authentication
>>
>> try using "export" instead of "set" or add it to the dial string directly
>> that calls it
>>
>> {sip_use_gateway=broadview}sofia/.......
>>
>>
>> On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <wchao at yahoo.com> wrote:
>> Where do you recommend I put the sip_use_gateway=broadview action?
>>
>> I have tried in the conf/dialplan/public.xml like so:
>>
>> <extension name="public_did_broadview">
>> <condition field="destination_number"
>> expression="^(12675379324|2675379324|12675379325|2675379325)$">
>> <action application="set" data="sip_use_gateway=broadview"/>
>> <action application="transfer" data="$1 XML default"/>
>> </condition>
>> </extension>
>>
>> I've also tried in conf/dialplan/extensions/7_inbound.xml (a file I
>> created that is pulled in via an include
>> pre-processor directive):
>>
>> <extension name="broadview_inbound_9325">
>> <condition field="destination_number"
>> expression="^12675379325|2675379325$">
>> <action application="set" data="sip_use_gateway=broadview"/>
>> <action application="transfer" data="1001"/>
>> </condition>
>> </extension>
>>
>> I have a gateway named broadview in conf/sip_profiles/external. In
>> both cases, I still get the following error on
>> the Freeswitch console:
>>
>> 2008-10-31 10:37:28 [ERR] sofia_reg.c:1089
>> sofia_reg_handle_sip_r_challenge() No Matching gateway found
>>
>> On Fri, 31 Oct 2008, Anthony Minessale wrote:
>>
>> Date: Fri, 31 Oct 2008 08:04:23 -0500
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Reply-To: freeswitch-users at lists.freeswitch.org
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking
>> authentication
>>
>> See what they said in the challenge?
>>
>> WWW-Authenticate: Digest
>> realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"
>>
>> Since this is a spontaneous challenge (which i think is somewhat silly
>> since it lets you talk on the phone for 40
>> minutes then makes you authenticate to hangup but *shrug*) FS does not
>> know which gateway to use for credentials.
>>
>> The realm they sent was SipLocal so FS is looking in its configuration for
>> a gateway with that name.
>> The 2nd thing it tries is the host from the To: header (64.115.128.6).
>> if there was a gateway with either of those
>> names,
>> it would find it.
>>
>> So try naming your gateway SipLocal or 64.115.128.6
>> or you can try setting the variable sip_use_gateway=<whatever> on the
>> channel which can give it a hint which
>> gateway to use.
>>
>>
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>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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>>
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>>
>>
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
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