[Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 10

Michael Collins msc at freeswitch.org
Fri May 3 23:14:30 MSD 2013 

Put a console debug log on pastebin.freeswitch.org so that we can see
what's going on. Also, see this page for some handy troubleshooting tips:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin

-MC


On Fri, May 3, 2013 at 5:20 AM, Navnath Sonavne
<navnath.sonavne at yahoo.com>wrote:

> Hi,
>
> As you said create new extensions for flex client and h323 endpoint(Ekiga
> softphone).
> I created two new extensions 1103.xml for flex client and 1104.xml for
> ekiga soft phone.
> I login using 1103 on flex client and make call to 1104.
> Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga
> phone rings and when i answer the
> call ,call terminates immediately by showing local user rejected call.
>
> In another case where I call from Ekiga phone(1104) to flex
> client(1103),it says person at 1104 is
> not available and says record your voice mail.In this case flex client did
> not get any incoming call.
>
>
> Here is my default.xml
>
> <extension name="Local_Extension" continue="true">
>       <condition field="destination_number" expression="^(1104)$">
>       <action application="export" data="dialed_extension=$1"/>
>       <action application="set"
> data="effective_caller_id_number=${dialed_extension}"/>
>       <!--action application="set" data="call_timeout=30"/-->
>       <!--action application="set" data="hangup_after_bridge=true"/-->
>       <action application="set" data="bypass_media=true"/>
>       <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>       </condition>
>
>       <condition field="destination_number" expression="^(1103)$">
>       <action application="export" data="dialed_extension=$1"/>
>       <action application="set"
> data="effective_caller_id_number=${dialed_extension}"/>
>       <!--action application="set" data="call_timeout=30"/-->
>       <!--action application="set" data="hangup_after_bridge=true"/-->
>       <action application="set" data="bypass_media=true"/>
>       <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>       </condition>
> </extension>
>
> Correct me if my dialstrings are wrong.
> Tell me if any other changes to made in other files also.
> How can I make both way calling ?Please help me its emergency , I have
> demo next week.
>
> -Navnath.
>
>
>
>
>
>   ------------------------------
>  *From:* "freeswitch-users-request at lists.freeswitch.org" <
> freeswitch-users-request at lists.freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, 2 May 2013 8:59 PM
> *Subject:* FreeSWITCH-users Digest, Vol 83, Issue 10
>
> ----- Forwarded Message -----
>
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> Today's Topics:
>
>   1. make ekiga to flex call (Navnath Sonavne)
>   2. Re: make ekiga to flex call (Brian Foster)
>   3. Re: Check if UA is still there? (mehroz)
>   4. Re: Check if UA is still there? (Ken Rice)
>   5. Re: Check if UA is still there? (mehroz)
>   6. Re: Check if UA is still there? (Michael Collins)
> Hi All,
>
> I am using flex client(at 192.168.9.165) given in freeswitch source
> to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via
> freeswitch server(at 192.168.8.41).
> I have two users registered on freeswitch with extension as 1100 and 1101
> in default context.
> I loged in using one of extension in flex client,then I dail another
> extension.
> After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have
> made changes
> in default.xml accordingly to forward calls to this ip.
> Here is My default.xml part :
>
> <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(\d{4})$">
>       <action application="export" data="dialed_extension=$1"/>
>       <action application="set"
> data="effective_caller_id_number=${dialed_extension}"/>
>       <action application="set" data="PEER_IP=192.168.8.27"/>
>       <action application="set" data="call_timeout=30"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="set" data="proxy_media=true"/>
>       <action application="bridge" data="h323/$1@${PEER_IP}"/>
>       </condition>
> </extension>
>
> Ekiga phone can answer the call and there is audio transmission from both
> end successfully.
>
> But this is one way calling i.e. from flex to ekiga.
> Now I want to call from Ekiga softphone to flex client.
> How to call flex client from ekiga softphone?
>
> Anybody please me on this issue.
>
> -Navnath.
>
>
>
>
> According to your dialplan you're forcing all calls that hit
> Local_Extension (any call with a destination number of 4 digits) to call
> your ekiga softphone. Your other problem lies in the fact that you should
> have different dialplans for different methods of calling unlike endpoints
> one for h232 one for flex.
> See below:
> -BDF
> On May 2, 2013 9:05 AM, "Navnath Sonavne" <navnath.sonavne at yahoo.com>
> wrote:
> >
> > Hi All,
> >
> > I am using flex client(at 192.168.9.165) given in freeswitch source
> > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via
> freeswitch server(at 192.168.8.41).
> > I have two users registered on freeswitch with extension as 1100 and
> 1101 in default context.
> > I loged in using one of extension in flex client,then I dail another
> extension.
> > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have
> made changes
> > in default.xml accordingly to forward calls to this ip.
> > Here is My default.xml part :
> >
> > <extension name="Local_Extension">
> >       <condition field="destination_number" expression="^(\d{4})$">
> >       <action application="export" data="dialed_extension=$1"/>
> >       <action application="set"
> data="effective_caller_id_number=${dialed_extension}"/>
> >       <action application="set" data="PEER_IP=192.168.8.27"/> <<Bad: you
> should never hardcode a specific endpoint IP if this dialplan should be
> dialing others as well
> >       <action application="set" data="call_timeout=30"/>
> >       <action application="set" data="hangup_after_bridge=true"/>
> >       <action application="set" data="proxy_media=true"/>
> >       <action application="bridge" data="h323/$1@${PEER_IP}"/> <<Bad:
> you set your dialplan to bridge the call to your softphone
> >       </condition>
> > </extension>
> >
> > Ekiga phone can answer the call and there is audio transmission from
> both end successfully.
> >
> > But this is one way calling i.e. from flex to ekiga.
> > Now I want to call from Ekiga softphone to flex client.
> > How to call flex client from ekiga softphone?
> >
> > Anybody please me on this issue.
> >
> > -Navnath.
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > 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
> >
> What if we want to even hungup the channels associated with those users?
> does
> freeswitch helps ?
>
> Like, i have a scenario in which , an active call disrupts , once BOTH
> users
> gets disconnected from network.
> Here, both users gets unregistered after option messages but the call
> sustains. Is there any thing i can handle this scenario?
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
> options pings are really just for registered users. a user does not have
> to be registered to place a call. more useful in your scenario is rtp
> timers and session timers
>
> Ken
> Sent from my iPad
>
> On May 2, 2013, at 8:49, mehroz <mehroz.ashraf85 at gmail.com> wrote:
>
> > What if we want to even hungup the channels associated with those users?
> does
> > freeswitch helps ?
> >
> > Like, i have a scenario in which , an active call disrupts , once BOTH
> users
> > gets disconnected from network.
> > Here, both users gets unregistered after option messages but the call
> > sustains. Is there any thing i can handle this scenario?
> >
> >
> >
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > 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
>
>
> i cannot use session-timers, disabled due to an issue with video
> calling....
>
> and rtp timers just does not seems to be working!
>
> i have
>
> But once network disconnects from both clients (works good, with a single
> client network outage), FS does not hangs up the call .....
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
>
>
> On Thu, May 2, 2013 at 8:13 AM, mehroz <mehroz.ashraf85 at gmail.com> wrote:
>
> i cannot use session-timers, disabled due to an issue with video
> calling....
>
> and rtp timers just does not seems to be working!
>
> Can you elaborate on this? Do you have a full pcap and FS debug log of a
> call with RTP in both directions and then the incoming RTP stream stopping
> and yet the call not being disconnected?
>
>
> i have
>
> But once network disconnects from both clients (works good, with a single
> client network outage), FS does not hangs up the call .....
>
> Correct. No SIP signaling and no session timers means there's nothing left
> to tell FS that the call should be torn down, except for the RTP timers
> mentioned above. That's why you'll need to supply a complete debug w/ pcap
> in order to find out why RTP timers are not working.
>
> -MC
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
>
>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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