October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hangup problem/SIP BYE lacking authentication
by Wellie Chao 10 Nov '08

10 Nov '08

Hangups do not work for me under certain circumstances. Here is the background information: * Our carrier uses a Metaswitch server with Acme Packet in front as a proxy/SBC. Only the Acme Packet machine is publicly visible (64.115.128.6). * Our Freeswitch server is at 216.57.23.143. * For calls originating from Freeswitch and terminating on Metaswitch (216.57.23.143 -> 64.115.128.6), Freeswitch authenticates with Metaswitch and everything works hunky-dorey. Either side can hang up and the other side will automatically hang up without requiring a manual hang-up. Now the problem: For calls originating from Metaswitch to Freeswitch (64.115.128.6 -> 216.57.23.143), Metaswitch does not authenticate with Freeswitch. Metaswitch also does not use the existing authenticated registration that our Freeswitch server initiates with Metaswitch upon startup of Freeswitch. Metaswitch just begins a new (unauthenticated) session and we have configured Freeswitch to allow any inbound calls from 64.115.128.6 without requiring authentication. We receive inbound calls (Metaswitch to Freeswitch, 64.115.128.6 -> 216.57.23.143) just fine. The phone rings and we can have a normal conversation. If the caller (the endpoint attached to Metaswitch) hangs up, both sides hang up. If I hang up (remember, I'm at the endpoint attached to Freeswitch), the caller's line remains attached forever. I have recorded a packet trace. Look at freeswitch_2.cap in the ZIP file, and you want to graph the first call starting at 21.202 and ending 53.798. If you go to time 53.087, you can see that my Freeswitch server sends a BYE to Metaswitch. This is a result of me hanging up my phone. At time 53.089, you see Metaswitch responding with 401 Unauthorized. Later at time 53.777, you see a BYE from Metaswitch to Freeswitch, but you should ignore this because that was a result of the caller (the guy hooked up to Metaswitch) manually hanging up. If he had not hung up his phone, the BYE from Metaswitch to Freeswitch would not have been issued and his phone would just stay on the line forever. Also, when I hang up my phone, I see the following at the Freeswitch console: 2008-10-29 23:03:28 [ERR] sofia_reg.c:1089 sofia_reg_handle_sip_r_challenge() No Matching gateway found I presume that Freeswitch emits this error because it got the 401 Unauthorized from Metaswitch. I also asked our carrier for a packet trace of a successful hangup on the Aastra platform (the engineer at the carrier says it is an Asterisk derivative -- I'm not sure about that). Look at Aastra_authentication_test.cap in the ZIP file. Graph the first call starting at 43.633 and ending 93.156. If you go to 93.118, you'll see that the Aastra server sends a BYE. Just like our Freeswitch scenario, Metaswitch sends back a 401 Unauthorized, but in response to the 401 Unauthorized, Aastra then sends back another BYE with the difference that the second BYE is authenticated. Metaswitch gets the second BYE and responds with 200 OK. I am pretty sure that if Freeswitch were to send back a second BYE (but with authentication), it would work fine. Now my question is how can I do this? I am not sure if this divergence of behavior is caused by: (a) my own error in configuring Freeswitch, (b) Metaswitch lacking standard SIP support (maybe it's not supposed to send the 401 Unauthorized), or (c) Freeswitch lacking standard SIP support (maybe it's supposed to send back a second BYE with authentication automatically). I don't know the SIP standards (or Freeswitch) well enough to know whether this problem is caused by me or by a deficiency in one of the two products (Metaswitch or Freeswitch). Can you provide some pointers? The ZIP file with the packet traces can be downloaded here: http://216.57.23.143/hangup_problem.zip

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What happend to variable_* in socket_outbound?
by Andy Spitzer 05 Nov '08

05 Nov '08

Woof! I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back? --Woof!

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Re: [Freeswitch-users] get channel status
by B Karthik 04 Nov '08

04 Nov '08

Set hangup_after_bridge variable to false and run the info command after bridge, you will see a lot of variables with the channel's current status. You could later reference those variables directly in your xml / script dialplan. ---------- Forwarded message ---------- > From: "Gopal krishnan" <saigop(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 29 Sep 2008 15:40:59 +0530 > Subject: [Freeswitch-users] get channel status > Hi, > > Is there any possibilities that I can check my channel status whether it > is ringing or answer or hangup. I am trying to fetch thru uuid but couldn't > able to do that. > > any suggestion would be helpful. thanks > > -- > Thank you with regards, > Gopal, > >

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lumenvox issue
by Frederick Jabre 01 Nov '08

01 Nov '08

Having trouble with Lumenvox. Could be a config error. I had this working with an older build. Mod_lumenvox is loaded successfully but the pizza demo keeps hanging up on me with this error: 2008-07-14 10:21:47 [NOTICE] switch_ivr_async.c:1773 switch_ivr_detect_speech() Hangup sofia/internal/fred-sip(a)200.56.122.8[CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-14 10:21:47 [NOTICE] switch_core_session.c:802 switch_core_session_thread() Session 1 (sofia/internal/fred-sip(a)200.56.122.8) Ended 2008-07-14 10:21:47 [NOTICE] switch_core_session.c:804 switch_core_session_thread() Close Channel sofia/internal/ fred-sip(a)200.56.122.8 [CS_HANGUP]

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How to calculate PDD for a call
by shehzad p 01 Nov '08

01 Nov '08

I want to calculate PDD (Post Dial Delay) for a call starting from when the call was originated from fs upto the time when the number starts ringing. (When fs receives 180 ringing from receiver side). I am originating call from js script. So how can I calculate PDD for a call. Any help would be appreciated. -- View this message in context: http://www.nabble.com/How-to-calculate-PDD-for-a-call-tp20110120p20110120.h… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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apr_md5 windows build problem
by Tamas Cseke 31 Oct '08

31 Oct '08

Hello, I have a problem with windows build: Microsoft Visual C++ 2005 Revision: 10158 Creating library Debug/FreeSwitchCore.lib and object Debug/FreeSwitchCore.exp switch_apr.obj : error LNK2019: unresolved external symbol __imp__apr_md5@12 referenced in function _switch_md5@12 C:\freeswitch\freeswitch-trunk\Debug/FreeSwitch.dll : fatal error LNK1120: 1 unresolved externals Build log was saved at "file://c:\freeswitch\freeswitch-trunk\w32\Library\Debug\BuildLog FreeSwitchCoreLib.htm" FreeSwitchCoreLib - 2 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 0 up-to-date, 0 skipped ========== Thank you, Tamas

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Help with Pocketsphinx and setting up pizza demo on Windows XP
by mszlazak＠aol.com 31 Oct '08

31 Oct '08

First, what version of pocketsphinx comes with the current FS? I understand that the latest release from Carnagie Mellon of pocketsphinx is version 0.5. Second, how do I set up the "pizza demo" with FS on Windows XP?? The wiki "Mod pocketsphinx" (http://wiki.freeswitch.org/wiki/Mod_pocketsphinx) seems only for Linux but I have unpacked and placed the sound and grammar files in similar folders on XP. I'm only missing the script file ps_pizza.js. Where do I get that? What else do I need to do to get this demo running on my system? Thanks. Mark.

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Problem with mod_openzap.c and SS7boost
by Helmut Kuper 31 Oct '08

31 Oct '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, we use FS with mod_openzap to talk to a ss7boost server. we found that when we cancel a call to PSTN befor media is established by FS the pstn side won't be canceled as well. pstn side rings until timeout or voicebox. I changed in mod_openzap.c in function "channel_on_hangup" line 446 form if (tech_pvt->zchan->state != ZAP_CHANNEL_STATE_DOWN && tech_pvt->zchan->state != ZAP_CHANNEL_STATE_TERMINATING) { to if (tech_pvt->zchan->state != ZAP_CHANNEL_STATE_TERMINATING) { That works so far. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkkG2+UACgkQ4tZeNddg3dzylACfXMJY/KU65+XTqqxUBYFmlK/2 lQcAn2CIsotR+cBW5R5S6YLyl9GXIt2K =26vq -----END PGP SIGNATURE-----

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Freeswitch Consulting
by Gerry Hull 31 Oct '08

31 Oct '08

I need a couple of hours of someone's time to resolve some FreeSwitch/SIP issues. Our application is on Windows. I've emailed consulting(a)freeswitch.org with no results. Please PM me if you are interested. Gerry

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Freeswitch as SBC ?
by Andreas Dimitriou 31 Oct '08

31 Oct '08

Hello, Has anyone used Freeswitch as SBC ? BR Andreas D

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