April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fs prepaid wholesale
by budi wibowo 26 Apr '11

26 Apr '11

hi any idea to implement prepaid wholesale for FS? thx budi

4 4

absolute_codec_string question
by Nicolas Brenner 26 Apr '11

26 Apr '11

Hi, I'm trying to force the use of a specific codec when originating a call from the command line. This is my codecs config in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,SPEEX,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,SPEEX,G729"/> If I just originate a call: originate {ignore_early_media}sofia/gateway/mygateway/444444 &bridge(user/1001) I get the following SDP: v=0 o=FreeSWITCH 1303727406 1303727407 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 17054 RTP/AVP 0 8 98 18 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 If I originate a call with the absolute_codec_string variable set like this: originate {ignore_early_media,absolute_codec_string='G729,PCMU'}sofia/gateway/ mygateway/444444 &bridge(user/1001) I get the following SDP: v=0 o=FreeSWITCH 1303719860 1303719861 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 24894 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 And it doesn't matter the value I give to the absolute_codec_string, I will get the same SDP, except if I specify SPEEX, then I'll get the same SDP as the one above. Am I missing something, or is this the way it's supposed to work? I also tried with the codec_string variable, and I get the same results. I'm using the following version from the git repository: head-git-244fd68 2011-03-21 14-27-57 -0400 Thanks for your help, Nicolas

7 14

continue_on_fail for condition only where 100 not received?
by peely 26 Apr '11

26 Apr '11

Hi, I can't seem to get a continue_on_fail code which would only retry if no response whatsoever is received from an endpoint. Does anyone know how I could route to a secondary endpoint ONLY if the initial endpoint does not respond with 100? I guess as a secondary part of the question I would need to know if I can change the INVITE timer to a smaller amount so the second device is attempted in less than 60 seconds. Basically I have a bunch of routing devices within the network I want to route to with a failove option, these devices onward route so I trust if that device is OK not to route to the second device, given these devices log sessions themselves it's vital that if one device receives an INVITE none of the others do for the same session. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/continue-on-fail-for-conditio… Sent from the freeswitch-users mailing list archive at Nabble.com.

6 8

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

2 1

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.

3 5

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 >>> >>> >> >

1 0

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:

1 0

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 >> >> >

1 0

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.

3 4

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.

3 4