- FreeSWITCH-users - FreeSWITCH Mailing Lists

answering machine detection
by Sam 26 Apr '11

26 Apr '11

Hello, Currently i see there is a module for voicemail detection where in it detects the beep, is there a module for detection of answering machine ( automated voice IVRS ), or a way to integrate to FS or a code to modify to detect automated voice or IVRs ? Regards Sam

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SIP router/switch only configuratoin advice?
by Georgy Goshin 26 Apr '11

26 Apr '11

Hello! I need some advice from experienced FreeSWITCH users. Please review my needs and advise me some configuration samples or adapted configuration guides. I'd like to use the FS in two roles - a SIP router for my PBXes and as a wholesale switch. Both configurations are almost the same in my case. The desired configuration should be allow me to do the following: route calls based on A num, B num, SRC IP (Mera MVTS is an example of what I want in simple calls routing). If a more complex configuration that can do dynamic routing based on price, acd/asr and prepaid/credit-limit billing is possible too, I'd be glad to see some examples too. Thanks in advance, G. Goshin.

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Re: [Freeswitch-users] was --> No Dial Tone ... Zaptel + FreeSwitch + TDM400P
by Pablo Silva 26 Apr '11

26 Apr '11

Dear Friends: This article is out? http://wiki.freeswitch.org/wiki/OpenZap_Dahdi, or is exactly to do for my case..? Greetings, Pablo On Tue, Apr 26, 2011 at 11:06 AM, Pablo Silva <psilvao(a)gmail.com> wrote: > Opss forget pastebin ... > > http://pastebin.com/87WRcsE2 > > On Tue, Apr 26, 2011 at 11:05 AM, Pablo Silva <psilvao(a)gmail.com> wrote: >> well.., now i get freetdm installed, the last version, but dosen't >> works.., i can't get dial tone and don't know why.. :-(, only i can >> see the message on my sip phone >> >> Service Unavailable >> >> >> >> That's my logs and configuration files, i'm using TDM400P with one FXO module .. >> >> Thanks in advance for any hint >> -Pablo >> >> Pd: I'm Chilean for these reason you can see in tone zone cl (Chile). >> >> >> On Mon, Apr 25, 2011 at 10:50 AM, Meftah Tayeb <tayeb.meftah(a)gmail.com> wrote: >>> openzap have bean renamed to freetdm >>> so please try mod_freetdm >>> thank you. >>> On 25/04/2011 14:56, Pablo Silva wrote: >>>> >>>> Hi! >>>> >>>> >>>> I've tryed for severals hours to make a phone call from freeswitch >>>> server, using TDM400P. >>>> >>>> I don't know why i can't hear the dial tone, and it's impossible >>>> make a phone call.., i read freeswitch wiki page >>>> about TDM400P configuration, as you can see that's my configuration an log >>>> >>>> http://pastebin.com/Y94UqN5R >>>> >>>> if I focus on information from the log, you can see that loads >>>> openzap, the server recognizes the TDM400P card, the call is routed to >>>> the PSTN but no concrete achievements. >>>> >>>> When you call, I hear the tone, not the dialing, I guess I'm near the >>>> solution to my problem, I request your cooperation. >>>> >>>> In advance, thanks. >>>> -Pablo >>>> >>>> _______________________________________________ >>>> 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 >>>> >>> >>> >>> -- >>> Meftah Tayeb >>> inum: +883510001288000 >>> phone: +13477595883 >>> >>> >> >

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Audiocodes phones
by Yehavi Bourvine 26 Apr '11

26 Apr '11

Audiocodes has announced recently two IP phones. (single line and 4 lines). I've updated the wiki at http://wiki.freeswitch.org/wiki/Interop_List#AudioCodes with more pointers from there. The phones are targetted toward Asterisk (as this is the PBX they use) but they are doing some changes to suit FreeSwitch as well. I am in touch with the product managers there, asking them for more features... Regards, __Yehavi:

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Re: [Freeswitch-users] No Dial Tone ... Zaptel + FreeSwitch + TDM400P
by Pablo Silva 26 Apr '11

26 Apr '11

Opss forget pastebin ... http://pastebin.com/87WRcsE2 On Tue, Apr 26, 2011 at 11:05 AM, Pablo Silva <psilvao(a)gmail.com> wrote: > well.., now i get freetdm installed, the last version, but dosen't > works.., i can't get dial tone and don't know why.. :-(, only i can > see the message on my sip phone > > Service Unavailable > > > > That's my logs and configuration files, i'm using TDM400P with one FXO module .. > > Thanks in advance for any hint > -Pablo > > Pd: I'm Chilean for these reason you can see in tone zone cl (Chile). > > > On Mon, Apr 25, 2011 at 10:50 AM, Meftah Tayeb <tayeb.meftah(a)gmail.com> wrote: >> openzap have bean renamed to freetdm >> so please try mod_freetdm >> thank you. >> On 25/04/2011 14:56, Pablo Silva wrote: >>> >>> Hi! >>> >>> >>> I've tryed for severals hours to make a phone call from freeswitch >>> server, using TDM400P. >>> >>> I don't know why i can't hear the dial tone, and it's impossible >>> make a phone call.., i read freeswitch wiki page >>> about TDM400P configuration, as you can see that's my configuration an log >>> >>> http://pastebin.com/Y94UqN5R >>> >>> if I focus on information from the log, you can see that loads >>> openzap, the server recognizes the TDM400P card, the call is routed to >>> the PSTN but no concrete achievements. >>> >>> When you call, I hear the tone, not the dialing, I guess I'm near the >>> solution to my problem, I request your cooperation. >>> >>> In advance, thanks. >>> -Pablo >>> >>> _______________________________________________ >>> 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 >>> >> >> >> -- >> Meftah Tayeb >> inum: +883510001288000 >> phone: +13477595883 >> >> >

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Receiving SMS some alert?
by Rodrigo Ferrari 26 Apr '11

26 Apr '11

Hello friends, I just bought a Khomp board to send and receive SMS, the problem is, I'm not receiving the messages, it's not marking at the console a message incoming. Is this some configuration that I miss? Thanks, best regards. Rodrigo Ferrari.

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mod_spandsp fails to cross compile
by mazilo 26 Apr '11

26 Apr '11

I don't remember when this problem started; however, I have been doing a git pull every time trying to re-compile my local FS source for the past two weeks and always ended up with the same error messages on mod_spandsp compilation when trying to cross compile FS as shown http://pastebin.com/7uKGPWxQ here on line #13 (mod_spandsp_fax.c:1656:13: error: implicit declaration of function 't38_gateway_rx_fillin'). ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-fails-to-cross-co… Sent from the freeswitch-users mailing list archive at Nabble.com.

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mod_nibblebill updates
by Rogelio Perez 26 Apr '11

26 Apr '11

Hello, I'm using mod_nibblebill to bill my calls only for leg b and heartbeats set to 'off'. After the hangup I see 3 different MySQL updates from nibblebill instead of just one. This is my dial plan: <extension name="TEST"> <condition field="destination_number" expression="^(1\d+)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/> <action application="hangup"/> </condition> </extension> ...this is the MySQL log: 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.343372 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.001177 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' 68080 Query select 1 68080 Query UPDATE web_account SET balance=balance-0.001197 WHERE id='13' 68080 Query select 1 68080 Query SELECT balance AS nibble_balance FROM web_account WHERE id='13' ...and here's the FS log: http://pastebin.freeswitch.org/16175 The xml_cdr is saved after the second update, so the nibble_total_billed is different than the real total value debited from the account balance. Is this normal? Shouldn't nibblebill just do one update after the hangup?

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call not connecting sometime
by Philippe Le Toquin 26 Apr '11

26 Apr '11

Hello, I had the problem a few weeks ago but after a reinstall (unrelated to that) the problem had gone so I put it done as my messing up my system at the time! But today I had the same issue again. When I call sometime the call is not going through completely The symptom on my side are that nothing happens (no ring tone) on the other they say that the phones rings but when they pick up the phone they can't hear anything. Below is a siptrace where I changed the number and IP so I hope I didn't mess it up too much. http://pastebin.freeswitch.org/16165 Can someone let me know if they see something wrong? I tried to understand but it is beyond me :( Regards /Philippe

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Re: [Freeswitch-users] how to pass arbitrary headers from A leg to B leg when bridging
by Dave Horton 26 Apr '11

26 Apr '11

So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now. But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity. First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do. I basically want to use FS as a simple transcoding server between two endpoints, call them A and B. Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex. The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration). Simple, right? The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header. So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg. So far, the only way I have found to do that is to hack the code to create a new channel variable. I've done this, and it works. However, this leads me to the following questions 1) Is there a better way to do this? If there is no way to do it as a dialplan out of the box, can it be done as a script? 2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia. Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)? Dave

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