June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS server and SPA3102 connection
by toddbailey 07 Jun '12

07 Jun '12

Hi All, My research leads be to believe I can use the Cisco SPA3102 as a device to connect the phone line to the fs server (via a lan cable) and I won't need a internal phone card installed. Am I correct in this belief or do I need something like a x100P card, a different phone card or compatible pci modem card to process incoming/outgoing phone calls to an analog phone service? thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-server-and-SPA3102-connect… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch is responding to contact's port when client behind NAT
by BELint Inc 07 Jun '12

07 Jun '12

Hi All, We`ve been using freeswitch since very long, It was good all over but it got stuck to a simple yet complex situation. We have Yealink T28P series phones behind NAT and freeswitch is unable to get them register. In the scenario freeswitch is at public IP. The reason we think is that freeswitch is using sip port in contact header to respond to. Is there anyway we can avoid this or force it to always send the response to destination IP instead of contact or via header ports because clients are behind NAT and response can only reach them if it respond to destination IP & PORT. Please revert if someone know how to solve this problem. It is very urgent. Regards

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RTP NAT issue
by Carlo Dimaggio 07 Jun '12

07 Jun '12

Hi all, I have a problem with RTP and NAT. The scenario is Hosted PBX and Natted phones (yealink): Phones (192.168.0.x) - NAT -> FS (public IP) When I call FS (for example an IVR) from a Phone, FS send the RTP to the private address (192.168.0.x) instead to the public NAT IP. The registration is ok: freeswitch@internal> sofia status profile tenant1.bs.dev.voip.clio.it reg 1 Registrations: ================================================================================================= Call-ID: 488014850(a)192.168.0.100 User: 202(a)tenant1.test.com Contact: "Test 202" <sip:202@192.168.0.100:5062;fs_nat=yes;fs_path=sip%3A202%40<NAT_IP>%3A37710> Agent: Yealink SIP-T20P 9.61.0.70 Status: Registered(UDP-NAT)(unknown) EXP(2012-06-06 19:01:55) EXPSECS(3232) Host: localhost.localdomain IP: <NAT_IP> Port: 37710 Auth-User: 202 Auth-Realm: tenant1.test.com MWI-Account: 202(a)tenant1.test.com How I can tell FS to send the RTP Packets to the right address? I think is needed a "comedia mode" like in Asterisk (or RTPProxy in openser)... Best regards,

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Re: [Freeswitch-users] Help me please
by Sixto Palacios 07 Jun '12

07 Jun '12

From: gmaruzz(a)gmail.com Date: Thu, 7 Jun 2012 18:00:38 +0200 Subject: Re: FW: Help me please To: syxtyn(a)hotmail.com Please send help requests to the mailing list. You can find the mailing list at: http://lists.freeswitch.org/mailman/listinfo/freeswitch-users On Thu, Jun 7, 2012 at 5:13 PM, Sixto Palacios <syxtyn(a)hotmail.com> wrote: From: syxtyn(a)hotmail.com To: gmaruzz(a)celliax.org Subject: Help me please Date: Thu, 7 Jun 2012 09:04:58 -0600 Hello friend, how're u?, need help with this error in freeswitch on Centos 6.2, please sofia.c:5000 Ping succeeded asterisk with code 200 - count -1/1/1, state UP 2012-06-07 08:55:07.500404 [ERR] mod_skypopen.c:1818 [07bc7ba|07bc7ba] [ERRORA 1818 ][skype101 ][IDLE,IDLE] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=voipufg, interface_id=1 FAILED to start. No Skype client logged in as 'voipufg' has been found. Please (re)launch a Skype client logged in as 'voipufg'. Skypopen exiting now Violación de segmento (`core' generado) -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Re: [Freeswitch-users] Skype on FreeSwitch install issues
by toddb＠toddbailey.net 07 Jun '12

07 Jun '12

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Re: [Freeswitch-users] Problems with multiple-registrations
by Alex Lake 07 Jun '12

07 Jun '12

OK - beginning to get to the bottom of this... The sofia_contact command WAS showing multiple registrations, it just that I (for some unknown reason) expected to see them on separate lines, whereas they're shown on the same line, comma delimited. My bad.... However, there is (or was!) another problem in that one of our FS boxes is running a slightly old version (FreeSWITCH Version 1.0.head (git-139bd07 2011-11-21 08-27-25 -0600)) in which it would appear that multiple-registrations really is/was broken. Might that be possible? Either way, I am intending to upgrade that box overnight. On 07/06/2012 15:15, Alex Lake wrote: > Trying to set up our Freeswitch boxes to support "a few" SIP > registrations against individual accounts. > > I've modified "internal.xml" to have <param > name="multiple-registrations" value="true"/> but it doesn't seem to > have had the desired effect. > > If I 2 SIP phones registering against the same user, I have never seen > more than one simultaneous registration (as evidenced by the > "sofia_contact" command). However (and this really spooks me!) if I > call the DDI associated with that account, sometimes both phones ring > (and accepting the call on either will cause the other to stop ringing). > >

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freeswitch and new version of sepstral
by covici＠ccs.covici.com 07 Jun '12

07 Jun '12

Hi. Does mod_cepstral work with the new version 6 cepstral voices -- I don't want to install and try it before I have some idea. Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Problems compiling FS with mod_esl
by Pete Kelly 07 Jun '12

07 Jun '12

Hi I am currently having some problems compiling freeswitch in lenny with mod_esl. I have done this: - checked out FS - ./bootstrap.sh, ./configure - cd libs/esl && make && make luamod (as per wiki docs) - uncomment mod_esl from modules.conf I then run make all, and this error is produced when it gets to mod_esl.so: Does anyone have any advice? making all mod_esl Creating mod_esl.so... gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or directory gcc: /usr/local/src/freeswitch_generic/libs/esl/libesl.so: No such file or directory gcc -I/usr/local/src/freeswitch_generic/libs/esl/src/include -I/usr/local/src/freeswitch_generic/libs/curl/include -I/usr/local/src/freeswitch_generic/src/include -I/usr/local/src/freeswitch_generic/src/include -I/usr/local/src/freeswitch_generic/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -shared -o .libs/mod_esl.so -shared -Wl,-x .libs/mod_esl.o /usr/local/src/freeswitch_generic/libs/esl/libesl.so /usr/local/src/freeswitch_generic/libs/esl/libesl.so -lm -lz /usr/local/src/freeswitch_generic/.libs/libfreeswitch.so -L/usr/local/src/freeswitch_generic/libs/esl -lesl -Wl,--rpath -Wl,/usr/local/freeswitch_generic/lib -Wl,--rpath -Wl,/usr/local/freeswitch_generic/mod make[5]: *** [mod_esl.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_esl-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2

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att_xfer and loopback
by Alex Lake 07 Jun '12

07 Jun '12

(Reconnecting to Mailing List) Hi guys - so I go for a break to celebrate 60 years of the queen and look what's here now! Some really useful bits here. Thanks to Avi for answering in my place. The reasons he cites for desiring to use loopback are correct. However, because our scripts are auto-generated (from a base spec in our own XML format), it is viable (if somwhat undesirable) to spew out a superficially long and complex lua script as an alternative. One of the (seemingly radical) things we're trying to do is to offer SIP-grade functionality to mixed SIP & PSTN endpoints. This is best illustrated with an example: User Fred is a mobile kind of guy - although he will generally take calls on the SIP handset on his desk, he is often out and about, and so will also take calls on his mobile. His company has one of our hosted pbx services and he's extension 305. He's set things up so that the calls ring first on his SIP handset and, after a delay of 5s, on his mobile. Because mobiles have voicemail systems that can get in the way, he has the "voicemail defeat" option, whereby on answering the call (and hearing the whisper) he has to press 1 to show it's really him accepting the call. A colleague, Bill, on extension 301 is in a very similar situation. When Bill and Fred are both in the office, things are easy: If Bill gets a call that he wishes to handover to Fred (with a briefing), he can do a simple attended transfer SIP to SIP. When they're both out of the office, things are a little more complex: Firstly, Bill takes the call on his mobile (which doesn't have a transfer button) - he uses the * * key sequence to access a mid-call menu, selects the "attended transfer" facility and keys in Fred's extension number (305). The attended transfer script must do just what a normal call to Fred's DDI would do - i.e. Call the SIP handset then, 5s later, start trying to find him on his mobile (with the "push 1 to accept" feature). Hopefully that gives an idea of how what we're doing is a bit complicated. I will definitely try Antony's runes and see how that works. I'm also up for trying to "improve" the wiki from the point of view of FreeSwitch beginners. The main thing I would suggest is much more use of examples (and fully contextualised examples, not just free-floating parameters!) Anyway, thanks all (and hope this email gets through to the list)

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how to: get session variable after api_hangup_hook
by cristian re 07 Jun '12

07 Jun '12

Hello, I write a custom cpp module for freeswitch that exposes an API: SWITCH_ADD_API(commands_api_interface, "my_hangup", "my_hangup", my_hangup_api , MY_HANGUP_USAGE); I want to call this api from dialplan (after bridge hangup) for reading the variable "billmsec". I put this into my dialplan: <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=my_hangup_api"/> After hangup freeswitch call correctly my API but I have not figured out how to get the variable.

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