similar to: AW: DTMF detection when outgoing call to mobilephones

Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there

Hiya all, I have had no end of trouble trying to get my A101 E1 card working on a new asterisk installation. The sangoma tech people have ignored my emails about this. All the installation of wanpipe seems to go ok, and zaptel. it all installed compiled and does all the wanpipe hwprobe exactly as documented. Asterisk compiled ok, but when I run it give me Jun 27 11:12:18 ERROR[10528]:

Hello, I am wondering if someone has got any ideas that can help solve this problem. I have a dial plan that you call into, and depending on certain conditions it calls out on a number grabbed from a database. Something like this : exten => s,n,Do something exten => s,n,Do something else exten => s,n,Dial(ZAP/g1/${OUTBOUND},${timeout}) I need to log the time the person

Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

Hi there. The title basicly explains it. When we get incomming calls from cellular phones, the callers tend to echo ALOT. They hear their own voice at very high volums. This is a problem only for mobilphone users that calls in to us. I'm using wifi IP-phones. Asteirsk: CVS-Nv1-0-7-04/19/05 Any way to fix this on the asterisk server? Regards, Arne Morten --------------

Qestion hvow can i use my SE w800i to reed flac files from a meorycard Best regards Fredrik

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

Hi all, I've setup a PBX in production environment last week but we have immediatly find out drop conversations and lot of errors (in asterisk logs) like: Apr 10 17:30:48 NOTICE[25154] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 My configuration is: Motherboard: SuperMicro P8SCT (Intel E7221 chipset) Hypertrading and APIC enabled CPU: P4 3GHz Digium TE205P

Hello ml, i need some help on my zaptel configuration. My E100P only shows some YELLOW / RED alarm when I load the wct1xxp module and do a cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS YELLOW RED ... .. . My /etc/zaptel.conf is: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nl defaultzone=nl I tried zaptel-1.0RC2 and the latest CVS

Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system. what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx. Thanks inadvance regards rudra -------------- next part -------------- An HTML attachment

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

Hi all, i am stuck with the configuration of asterisk - modules are loaded ( zaptel and wct4xxp ) - i have zaptel.conf configure, output of ztcfg -vv --- snip -- rapid:~# ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hi, I am having problems with an Asterisk with a Digium TE205P card. The issue is that the Ring Back Tone is noisy. I am making modem's calls and this noise influences on the initial negotiation protocol, so modems have to recall. My configuration is: Asterisk version: Asterisk 1.4.21.2 Linux version: CentOS release 5.2 (Final) Card: Digium TE205P