August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Comparison matirx
by Simon Shaw 03 Aug '08

03 Aug '08

I have built a comparison matrix that will enable me to compare the leading Open Source soft switches. In the attached spread sheet there is a first suggestion of what I think should be compared and relevant weightings and I would be glad to hear your feedback. Regards, Simon

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How to send a message from FS to SMS (?)
by Henk Oegema 03 Aug '08

03 Aug '08

In * I use the following to send a SMS message to my mobile phone, in case of no answer: [macro-send_sms_message] exten => s,1,NoOp(${ARG1}) exten => s,n,System(/usr/bin/curl -s "http://localhost/asterisk/sms/sms.php?number=${ARG1}&msg=At%20${ARG3}%20${A….") exten => 1200,1,Dial(SIP/1200,10,tT) exten => 1200,n,Macro(send_sms_message,00324763788xx ,${CALLERID(num)},${STRFTIME(${EPOCH},,%d.%m.%Y-%H:%M)}) ..... ..... This send f.e. the following message: At 03.08.2008-17:52 00313446413xx called. How can I do something simular in FS? Grds Henk

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Inbound gateways
by Simon Shaw 03 Aug '08

03 Aug '08

I would like to receive calls from an * box that is configured to trunk all calls starting with the digit 4 to my FS box. What happens now is that incoming calls are rejected with a 407, I set the param auth-calls to false for public DID numbers as shown below but this did not help. <extension name="public_did"> <condition field="destination_number" expression="^4(\d{4})$"> <param name="auth-calls" value="false"/> <action application="transfer" data="$1 XML default"/> </condition> </extension>

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Freeswitch as a pseudo-proxy
by Robert Dyck 03 Aug '08

03 Aug '08

I read that FS is a B2BUA and not a proxy. However a B2BUA can often behave as a call-stateful proxy. SIP INVITE comes to mind. Can FS passthrough other SIP requests? Can it mangle the headers to traverse NAT or hide the toplology?

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FS webapi vs api question
by Michael S Collins 01 Aug '08

01 Aug '08

Could someone please confirm this behavior and comment on whether it is expected or not? Using the api or webapi from mod_xml_rpc, one can fetch CLI commands. Some commands seem to work well using either method and others do not. Try these two commands and tell me what you see: http://fs.ip:8080/webapi/sofia?profile internal http://fs.ip:8080/api/sofia?profile internal Normally the webapi wraps the return data or adds a content type header to make it browser friendly. The api just dumps the data raw. However there are cases where api returns partial or no data. Another example is the command oz?list. It works fine with webapi but returns nothing with api. If you know the scoop on this could you please enlighten me? Thanks! -MC Sent from my iPhone

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CDR: meaning of date and time variables
by Birgit Arkesteijn 01 Aug '08

01 Aug '08

Hi all, I'm trying to understand the meaning (and difference) between some of the date and time variables that FreeSWITCH posts as CDR, using xml_cdr. I'm interested in particular in the variables below: **** Variables in Date/Time stamp: - start (as in start_stamp, start_epoch and start_uepoch) - - profile_start - - answer - stamp when call was answered - progress - - end - stamp when call was terminated **** Variables in Seconds, Milliseconds, Microseconds: - duration - bill - progress - progress_media - flow_bill I couldn't find any documentation on the data. If there is, could you please post the link? Otherwise is there anyone who could explain what the meaning is of these variables? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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DOCUMENTATION: "mod of the week"
by Michael S Collins 01 Aug '08

01 Aug '08

Gang, On our weekly conf call Ray (intralanman) had a great idea: to help get the default configs documented we are going to have a "mod of the week" and invite all community members to assist with the documentation process. To help out you can do any or all of the following: Write the wiki page :) Send questions about the topic to the users list - this will help us determine what information needs to be included in the documentation Send us any tips/tricks/traps you may be aware of. Those little nuggets of wisdom can help thousands of potential users. We are also open to suggestions. This week's mod is mod_cdr_csv. Please submit anything you can and we will get cracking. -MC Sent from my iPhone

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Call Forwarding not working properly with local extensions
by unknown 01 Aug '08

01 Aug '08

Hi there, does any one know how to tweak the default dialplan for the local extensions so that the SIP endpoints with call forwarding setup can work properly? My existing Aastra 9133i phone and Polycom IP320's call forwarding was working as expected when working with Asterisk, but when working with freeswitch, the call forwarding just not working, it always goes to voicemail. I have tested this with both 1.0.1 release, and the latest SVN trunk code. Thanks for your advice. Chris Dialplan for the local extensions: <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ext}"/>  <anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <anti-action application="set" data="transfer_ringback=${us-ring}"/> <anti-action application="set" data="call_timeout=30"/>  <anti-action application="set" data="hangup_after_bridge=true"/>  <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/> <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/> <anti-action application="answer"/>  <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension>

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FS returning 503 Maximum Calls In Progress
by Ron 01 Aug '08

01 Aug '08

Hello, Can someone please tell me all the places in the config files where the number of concurrent calls can be limited? I am doing SIP-SIP calls with media and everything is working fine, except that at about 260 call legs, the above SIP response is returned to the originating GW. Thanks in advance, Ron

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Re: [Freeswitch-users] Complete seperation of VOIP app from FS
by Erol Akarsu 01 Aug '08

01 Aug '08

Jonas, This is what I want. Do you have any documentation on mod_ivr_socket and how its API looks like? Thanks Erol Akarsu ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, August 1, 2008 9:46:30 AM Subject: Re: [Freeswitch-users] Complete seperation of VOIP app from FS I've started to do a mod_ivr_socket a long time ago, which I never completed. It should be considered as an alternative to the scripting languages and gives you an easier interface than the event socket. I created it since I wanted to move the ivr applications from my javascripts into my own server coded in c#. I haven't had time to finish it though. I can add it as a contrib project if you're interested in completing it. On Fri, Aug 1, 2008 at 3:24 PM, Erol Akarsu <eakarsu_33(a)yahoo.com> wrote: > David, > > Thanks for your clarification. > > Here we have to write all these commands to send to FS remotely. > Do we have any high-level API that we can use to send command to FS and parse the results? > > Thanks > > Erol Akarsu > > > > ----- Original Message ---- > From: David Knell <dave(a)3c.co.uk> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Friday, August 1, 2008 12:19:49 AM > Subject: Re: [Freeswitch-users] Complete seperation of VOIP app from FS > > Hi Erol, > > A dialplan extension like this: > > <extension name="callthrough"> > <condition field="destination_number" expression="^0"> > <action application="socket" data="127.0.0.1:6100 async full" /> > </condition> > </extension> > > results in all calls to numbers starting with 0 being handled by a > server on localhost > listening on port 6100. That server has full control over the call. > Change the IP > address to run the server on a different host. > > Cheers -- > > Dave > >> Still searching event_socket module. >> If we set up dialplan, all received calls will routed based on dialplan rules. >> But at the same time, if there is remote process connected to FS through socket_module, dialplan or remotes process will have priority to control FS behavior. Can remote process interfere dialplan rules.? >> >> Is it possible to define a dialplan so that one extension will be routed to a remote process and other extension will be routed to different process? >> >> I am thinking to abstract out IVR or any VOIP application from FS box completely and to run application on different host or process. >> >> Regards >> >> Erol Akarsu >> >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > -- > David Knell, Director, 3C Limited > T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 > http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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