similar to: Very bad TDMF tone !

hello, asteriskers: I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run ./configure and menuselect with embedded modules. but running make comes out errors: ranlib libmxml.a make[3]: Leaving directory `/usr/src/asterisk-1.4.2/menuselect/mxml' cc -Wall -o menuselect.o -g -c -D_GNU_SOURCE menuselect.c cc -Wall -o menuselect_curses.o -g -c -D_GNU_SOURCE

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension => 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension => 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

hello, asteriskers: i compile asterisk 1.2.18 in arm-linux. i got this error :dlfcn.c:40: mach-o/dyld.h: No such file or directory. i check the /usr/include dir, there is no mach-o dir and dyld.h file in /usr/include. i think i am missing somethings in the cross-compile tools. Does nayone know that problem? please give me a hint! thanks! zhulizhong ---------------------------------

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you

I have a TDM400P card all installed, and I can call out from a sip phone, but when I call from the PSTN into the asterisk machine, as soon as the Answer() gets called, the dial tone changes and is sounds like there is a lot of static on the line. Below is the part of the dial plan for answering the call. exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Dial(Sip/pfriedel,20,tT)

New system install. At what point, in bootup, should I be able to get a dial tone on the phone ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug into <g>. At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not get a dial tone on the phone ports. I thought as long as zapata.conf is correct, the board should be

Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at

Hello, everyone. I'm having a strange problem with my asterisk. After dialing and before the other side picks up the phone I should hear the tones (I'm not sure what are they called: piiiiiiiiiiii---piiiiiiiiiiiiii....) and in almost all cases that is true. However there is a range of numbers where I'm having this problem. There is no tones, just silence, until someone picks up the

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, --

hello: thanks for Tzafrir Cohen for dahdi testing. I installed dahdi-2.1-r3c svn code and asterisk1-6 for testing OpenVox B400P and junghans card. i fund that there is bug (i think) to dectect NT or TE mode. actually on the board, i set it as TE mode, but after start wcb4xxp, but it show the port is NT mode. to detect the TE mode, I modefy the code in base.c

Hi List, I have a client who has a TDM400P with 4 FXO. He has a problem them when some one calls, then hangs up it takes a good 10-15 seconds or more of the card to realize that the line was hung up on. The phones keep reigning After a bit it hangs up on the line. Also there has been some hanging. (After a user on the PBX side hangs up the card does not "release the line"). I am using

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

Hello, I'm having trouble working out how to send DTMF tones to an external IVR. My system has an analog phone connected to a TDM400P card, a SIP software phone (Zultys LIPZ4) and is connected to a BRI in Australia with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched with the ISDN audio patch from Traverse (which allows the card to do voice). DTMF works fine between

Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or Cisco, etc), vs. cheap analog touch-tone phones, exactly what features will I kiss goodbye if I use the cheap

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line