similar to: false ringback

Index: channels/chan_alsa.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_alsa.c,v retrieving revision 1.2 diff -r1.2 chan_alsa.c 1042c1042 < if ((cfg = ast_load(config)) { --- > if ((cfg = ast_load(config))) { -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/

is there any particular reason why there is no option to configure the codec framesizes in iax2 ? It would come rathrer handy to decide if you want less bandwidth or more robustness on the payload side ... -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is

Matteo Brancaleoni wrote: >Hi. > >I'm experiencing a strange issue with *. >I have a dev kit, aka a T100P + a zhone cb. > >Sometimes, on certains phones (on the fxo ports >of the cb) , when the phone rings, * detect >it as answered after the first ring, even >if no one is at the phone! > >The result is that on the other party (which >called the phone) hears

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

Dear colleague, I'd like to compute linear discriminant analysis, using R. In the book Modern applied statistic with Splus (Venables & Ripley, p. 396), lda function is used. Could you tell me where I can find this function? At what site, can I download this library ? Thank for your help. Best Regards Sovan ---------------------------------------------------------------- Prof. Sovan

This was published on O'Reilly's site last week, but they didn't tell me until now. :) The article is pretty minimal, because I had a limited number of words to work with, so many features are not implemented. However, it's a good start. http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html I had a few people review it beforehand, but I'd also welcome changes or

Hi folks: Does the TDM400 card from Digium only support FXS, or is FXO functionality available or planned? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming, Networking

Anyone tested this card in Greece? I would like to buy several of these but I don't know if it works well with the PSTN in Greece. The Quicknet Linejack card seems to be unable to understand the telephone network's tones (busy, ringing,...). TIA, Michael.

Hello all, A newer version asterisk-oh323 (v0.5.5) is available. This version contains updates for compilation with latest Asterisk sources, some additions ("atexit" cleanup) and some other fixes. http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf:

If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 "session in progress"), so I guess I should be debugging the RTP packets. From then on

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

I am trying to get Asterisk to emulate the sounds of the earlier telephone systems, and the settings in [us-old] are pretty helpful. The only thing lacking is ringback tone, which is not quite as complex as the real phone systems of the day. For example, it is true that a ringback tone commonly used is 420Hz modulated by 40Hz. This is what shows up in [us-old]. But that modulated tone was

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in