January 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Removing inband DTMF
by Tamas Jalsovszky 17 Jan '11

17 Jan '11

Hello, Is there a way to remove an inband DTMF signal from the RTP stream with FreeSWITCH? We have a partner with ugly Cirpack which can not remove inband DTMF when the codec is g711. The problem is, that our other party wants RFC2833 only - thus we have to remove inband dtmf. Unfortunately Cirpack is not open to do this. Any advice? Kind regards, T.

6 8

Processing Live Audio during a call between 2 extensions
by Michel Habib 17 Jan '11

17 Jan '11

Dear Freeswitch Developers, i am relatively new to freeswitch and i am seeking some advice. I am creating an External Application that does some [live] processing on the Audio of the freeswitch call before sending it to the second Call Leg, and vice versa. Can you please direct me to the best way to do that and how to access/capture the live audio channel in both legs while it is streaming, and how to resubmit it again to the other leg? I read the book about freeswitch, but still couldnt figure the best approach to do that, any hint is appreciated. Best Regards, Michel Habib.

3 2

Codecs issue
by Diego Viola 17 Jan '11

17 Jan '11

Hello, I'm experiencing some strange issue with codecs. I have the following in my vars.xml file: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/> "inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode. However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters. I'm sending calls like this: <action application="bridge" data="sofia/internal/$1(a)38.102.93.70"/> Here is a SIP trace of a call: http://pastebin.freeswitch.org/15042 I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291. I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs? my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126 calls flow like this: softphone -> FS -> provider Any help appreciated.

5 18

core dump while using Codec2
by Meftah Tayeb 16 Jan '11

16 Jan '11

guys, here is a core dump while testing codec 2 module http://pastebin.freeswitch.org/15043 thanks -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313

2 1

SOFIA_REFER_TO_VARIABLE
by Madovsky 16 Jan '11

16 Jan '11

can anyone what the rul of this var ? Thanks

2 1

BLIND_TRANFER dialplan
by Madovsky 16 Jan '11

16 Jan '11

is it possible to create a blind transfer manually without deflect ? Thanks

2 1

Re: [Freeswitch-users] bridge a call to remote conference
by Madovsky 16 Jan '11

16 Jan '11

more weird, if I choose a codec from the sip phone based on 8000hz it doesn't work. it's like a rate conversion has to be forced to make the audio work on the conference bridge... ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Sunday, January 16, 2011 12:10 PM Subject: Re: [Freeswitch-users] bridge a call to remote conference after some test I confirm that it should force to offer a codec at the same rate of the conference, example <action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/> as GSM is a 8000hz code so it works now... thanks ----- Original Message ----- From: Steven Ayre To: FreeSWITCH Users Help Sent: Sunday, January 16, 2011 4:20 AM Subject: Re: [Freeswitch-users] bridge a call to remote conference Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep -Steve On 16 January 2011 03:54, Madovsky <infos(a)madovsky.org> wrote: everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec) but the hold music (in 8000hz) works well. should I set any special codec in the legB conference bridge ? Thanks _______________________________________________ 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 ---------------------------------------------------------------------------- _______________________________________________ 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

1 0

bridge a call to remote conference
by Madovsky 16 Jan '11

16 Jan '11

everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec) but the hold music (in 8000hz) works well. should I set any special codec in the legB conference bridge ? Thanks

2 3

Re: [Freeswitch-users] FS as sip proxy registration
by Peter Olsson 15 Jan '11

15 Jan '11

FS is not a proxy... /Peter ----- Reply message ----- Från: "Madovsky" <infos(a)madovsky.org> Datum: fre, jan 14, 2011 18:24 Rubrik: [Freeswitch-users] FS as sip proxy registration Till: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> if I use FS as SIP proxy registrar, does the REFER method will go through the client transparently ? I'm facing problem of this method with this configuration Thanks !DSPAM:4d30866d32761881510302!

3 2

mod_lcr and extra_vars
by Boris Kovalenko 15 Jan '11

15 Jan '11

Hello! I need to set extra vars with mod_lcr. I did as wiki recomended: 1) created sql column 2) modified sql quer 3) added <param name="export_fields" value="lcr_gw_extra_vars"/> to profile So, may lcr output looks nice: | Digit Match | Carrier | Rate | Codec | CID Regexp | Limit | Dialstring | | 734353 | tagnet.ru | 0.00000 | | | | [lcr_carrier=tagnet.ru,lcr_rate=0.00000,lcr_gw_extra_vars=,v_tagnet_ats_dstport=50000]sofia/epbx/73435350101@X.X.X.X:5060 Unfortunatelly I can't see (and can't use) v_tagnet_ats_dstport variable in my cdr records (even with b-leg the variable isn't present) until I set import=v_tagnet_ats_dstport variable. But I thinked mod_lcr should do this? Something wrong with my configuration? -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

2 8