[Freeswitch-users] No NOTIFY MWI when registering via proxy.
Peter P GMX
Prometheus001 at gmx.net
Sat Dec 5 12:35:28 PST 2009
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql> select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+
| sip_host | presence_hosts | server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
<document type="freeswitch/xml">
<section name="directory">
<domain name="sip1.mydomain.com">
<user id="100">
<params>
<param name="password" value="pass"/>
<param name="vm-password" value="pass"/>
<param name="vm-email-all-messages" value="true"/>
<param name="vm-attach-file" value="true"/>
<param name="vm-mailto" value="email at domain.net"/>
<param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
<param name="http-allowed-api" value="voicemail"/>
</params>
<variables>
<variable name="accountcode" value="800"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension 100"/>
<variable name="effective_caller_id_number" value="100"/>
</variables>
</user>
</domain>
</section>
</document>
The internal profile has the following alias:
<profile name="internal" domain="$${domain}">
<aliases>
<alias name="$${domain}"/>
<alias name="sip1.mydomain.com"/>
<alias name="default"/>
</aliases>
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
> based on your example past
>
> sip1.mydomain.com <http://sip1.mydomain.com> is the domain in the
> packet and thus the profile should have an alias for this.
> Then the user must reside in your sip db with the user 200 and domain
> sip1.mydomain.com <http://sip1.mydomain.com>
>
> if you dont have this consider the force-register-domain and
> force-register-db-domain to normalize the host names.
>
>
> On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
> Did you check the 2 replies that told you you need aliases in your
> sofia profile to translate the domain found in your
> message_waiting to the right profile? Both Brian and Mike
> answered you.
>
>
>
>
>
> On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
> <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>> wrote:
>
> I tried now with phones directly attached to the freeswitch
> (without an
> OpenSIPS in between). I also added the alias. But the
> behaviour is as
> before:
> No notify message from freeswitch, neither after register nor
> after a
> voicemail is recorded.
>
> Best regards
> Peter
> Brian West schrieb:
> > Yes an alias will be required for every domain you run on
> the profile
> > so it can find it.
> >
> > /b
> >
> > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
> >
> >
> >> Try an alias on the sip profile.
> >>
> >> Mike
> >>
> >
> >
> > _______________________________________________
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> >
>
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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> AIM: anthm
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> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest@conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest@conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
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