June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Weekly FreeSwitch Conference call June 9 @ 1pm EST
by Norman Tomlins 10 Jun '10

10 Jun '10

Weekly FreeSwitch Conference call June 9 @ 1pm EST Hi Everyone, I would just like to remind everyone that we have our weekly conference call on June , 9th @ 1pm EST. During this conference call Darren Schreiber will talk about mod_nibblebill. Information on mod_nilbble can be found here: http://wiki.freeswitch.org/wiki/Mod_nibblebill Information on Weekly Conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 Norman Tomlins ---------------------------------------------------------------------------- Voice Network Inc. http://www.VoiceNetwork.ca Unlimited Incoming US DID's - $ 3.95 - No Setup Fees Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees ----------------------------------------------------------------------------

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Callback with bypass media
by Vitalii Colosov 10 Jun '10

10 Jun '10

Hi, Could you please advice if this callback scenario can be achieved? Freeswitch calls one number via external provider. After it answered, it gets the digits and dials another number via external provider. After second phone is answered, it bridge both sessions. Is it possible that from this moment media will flow directly from one sip phone to another, bypassing freeswitch? And if yes, which command (in Lua preferably)? I tried to use the following code after getting the digits from first phone: ... calling_session:setVariable("bypass_media", "true"); called_session = freeswitch.Session(some_called_string, calling_session); ... But as per log, the INVITE to the second phone contains FreeSWITCH's IP address in SDP section, and not first phone's IP address. Looks like bypass_media is not working as I would expect in this case. Thanks, Vitalie

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Re: [Freeswitch-users] uuid_media hangs
by Kristian Kielhofner 10 Jun '10

10 Jun '10

That's completely ridiculous. -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: freeswitch-users(a)lists.freeswitch.org < freeswitch-users(a)lists.freeswitch.org> *Sent*: Thu Jun 10 12:30:42 2010 *Subject*: Re: [Freeswitch-users] uuid_media hangs > are very keen to actually follow all these silly RFC I agree, usually RFC are created first by university so by students... ----- Original Message ----- *From:* Anthony Minessale <anthony.minessale(a)gmail.com> *To:* freeswitch-users(a)lists.freeswitch.org *Sent:* Thursday, June 10, 2010 12:09 PM *Subject:* Re: [Freeswitch-users] uuid_media hangs The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely <freeswitch(a)peely.com> wrote: > > Hi, > > It's not a hijacked thread, I created this with an issue around uuid_media, > initially it was hanging, then in the latest git it receives a 4xx from the > far end due to the Requires: timer support bot being supported in the > reinvite transaction. I did anecdotally mention problems with events, which > I subsequently pulled off to another thread. > > I was looking at RFC 4028, in the examples it shows the refresh event not > providing the Required: timer header even for the refresh of the original > invite transaction, just the Supported: timer. There's nothing in RFC 3725 > regarding sesison timers in 3PCC but as the reinvite for the media > adjustment is a new transaction, I don't think it needs the Requires: timer > option to persist the timer support on the original dialogue. This is > certainly the behaviour of the previous equipment I had on this > interconnect > and it operates using session timers and 3PCC in several other environments > too. > > I'm not brilliant with C, but if you could please point me to the right > approach to removing the Requires: timer field in the header I'd be happy > to > test it and let you know if it causes any problems with rienvites across > various carriers and other platforms. > > > > Cheers, > > > Neil. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p51… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900 ------------------------------ _______________________________________________ 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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LUA Callback - Help
by Aloysius Lloyd 10 Jun '10

10 Jun '10

Hi All, I am trying to build a callback application using LUA. I have the following setup. - Dial the DID number - Hangup - Callback lua initiate the call after 10 secods The above working great. But when I answer the call ...the wav file never play on the channel.What I am doing wrong? <extension name="CallBack"> <condition field="destination_number" expression="^6471231234$"> <action application="set" data="api_hangup_hook=luarun callback.lua ${caller_id_number}"/> <action application="hangup" /> </condition> </extension> *callback.lua* os.execute("sleep " .. tonumber(10)); pin_number_test="0511"; out_gateway1="sofia/gateway/voipms/1"; message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; number_to_call= argv[1]; session = freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); session:setAutoHangup(false); if ( session:ready() ) then session:sleep(3000); session:streamFile(message_dir.."enter-dest.wav"); end Thanks Lloyd

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Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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LUA - session:waitForAnswer - Error
by Aloysius Lloyd 10 Jun '10

10 Jun '10

Hi All, In javascript the session.waitForAnswer(10000); working without any problem. But in Lua session:waitForAnswer(10000); throwing the following error. *2010-06-10 10:54:21.029607 [ERR] mod_lua.cpp:182 Error in waitForAnswer (arg 2), expected 'CoreSession *' got 'number' stack traceback:* What is the correct argument for this function in LUA. In wiki this function description is not available. Any help appreciate. Thanks Lloyd

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Problem witg message-waiting NOTIFY for Audiocodes ATA
by Yehavi Bourvine 10 Jun '10

10 Jun '10

Hello, Recently I've noticed that phones on our Audiocodes MP-124 adapter do not get the message waiting indicator. It seems like this behaviour has started a while ago after some upgrade (don't know whether it was Freeswitch or Audiocodes). The current versions are FS 1.0.6 and AudioCodes 5.6 >From various tests I've done with SIPSACK and tcpdumps I find the following: - after a Register command a "correct" notify is sent (i.e., the AudioCodes understands it). - later on "incorrect" notify's are sent which are rejected by the AC with "user not found". The "correct" notify is: NOTIFY sip:80633@10.64.0.8:5060 SIP/2.0 Via: SIP/2.0/UDP 10.64.0.3;rport;branch=z9hG4bKap2H5DSe6Zm7a Max-Forwards: 70 From: <sip:80633@pbx-gr.XXX>;tag=vtgH30805ZH1Q To: <sip:80633@pbx-gr.XXX>;tag=1c1710947583 Call-ID: 17109472461062010161640(a)10.64.0.8 CSeq: 131971366 NOTIFY Contact: <sip:80633@10.64.0.3:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=7200 Content-Type: application/simple-message-summary Content-Length: 106 Messages-Waiting: no Message-Account: sip:80633@pbx-gr.XXX Voice-Message: 0/0 (0/0) While the "incorrect one" differs in the following two lines: From: <sip:80633@10.64.0.3 <sip%3A80633(a)10.64.0.3>>;tag=vtgH30805ZH1Q To: <sip:80633@10.64.0.3 <sip%3A80633(a)10.64.0.3>>;tag=1c1710947583 Taking the second one, changing the domain name and sending it with SIPSACK made it work correctly. (10.64.0.3 is the adderss of pbx-gr.XXX and 10.64.0.8 is the Audiocodes). Whose side is not working properly and should be asked to be fixed? Thanks! __Yehavi:

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Freeswitch in an existing phone network
by Oliver Schenk 10 Jun '10

10 Jun '10

Hi All, The company I work for currently has quite an extensive phone network which gets carried between old analogue PABXes which also has an interface to IP based phones. All the phones in our office are connected via CAT5 cable using IP, however literally hundreds of phones out in the field (we operate railway infrastructure) are on standard voice analogue phones carried through fibreoptics. Anyway, I would like to use Freeswitch purely for its IVR and TTS abilities and nothing else. So basically I just need it to act like a slave to whatever IP phone network is already out there. I'm a bit worried if I fire up freeswitch it will hijack the phone network! All our phones are accessible via a 5 digit extension. I would like Freeswitch to be behind one of those ... say 12345. If anyone within our phone network dials 12345 then Freeswitch should answer. I guess my question is...how should I go about disabling most of FreeSwitch except it's ability to pick up the phone and speak IVR/TTS and make an outgoing call via the existing phone network? Any general pointers appreciated. Thanks, Oliver Schenk

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Gateway Registration
by Samuel Macedo 10 Jun '10

10 Jun '10

Hi, I want to use the same Gateway but with different users and passwords to make and receive calls. This is the scenario: I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents (Sofia External) to this gateway so I can receive inbound calls. And to make a Outbound call I want to decide witch UserAgent I will use. Is there anyway to do this? Regards, -- Samuel Macedo Belo Horizonte - Brazil

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Call Forwarding/Diversion Dialplan
by Michael De Lorenzo 09 Jun '10

09 Jun '10

I'm trying to write a dialplan that will forward a call to a PSTN number. PSTN (Party A) Inbound call to your Freeswitch (Party B), which is forwarded out to the PSTN (Party C) I've found examples for transferring calls to other Freeswitch users, but I haven't been able to get this to work properly--where it sets a Diversion header. Can anyone help?

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