similar to: Zap DTMF detection

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the "soft hangup Zap/<zapchannel>" in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1

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

I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local dialing. I launch a call "Zap/1/7705551212" and it goes thru just fine. The next time I try it, without any modifications, I get a Bell recording telling me that I must dial the area code and seven digit number when placing a local call. It's like Asterisk may be starting the dial before the line

I am going to try to figure out why mu asterisk box connected by back to back cable to an PRI appliance is not going to send the PROGRESSING dss1 message. In fact i see the SETUP and the follwing CALL PROCEEDING but not the PROGRESSING so the appliance doesn't allow the "early audio" !!!!! this is my zapata.conf [channels] context = ser switchtype = euroisdn usecallerid = yes

hello, We encountered signaling problem with a french national carrier. They ask us, which signaling is configured on our single E1. I need to know if it's ETSI, VN4 or VN6. I know what ccs, and hdb3 mean but I do not succeed to make the link between the signaling type. I searched through RFC Q.921 and Q.931 It would be great to obtain some help. cedric

Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages: *CLI> Warning, flexibel rate not heavily tested! Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2 Channel 4 unblocked Rx CAS bits 0x9 [ 10000/ 0/ 0] Line unblocked -- R2

Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Hello! I am trying to install a TN405P on a P4-3GHz-HT machine running Debian Sarge with kernel 2.4.27. When I start Asterisk in -vvvvc mode it always shows == D-Channel on span 1 up == Restart on requested on entire span 1 == D-Channel on span 3 up == D-Channel on span 2 up == Restart on requested on entire span 3 == Restart on requested on entire span 2 == D-Channel on span 4 up == Restart

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

Hi List, a number of our customers are reporting dropped calls. here is the config. 1 T100P T1 Card 1 Asterisk (Mid Nov build) T1 is signalled as a PRI(National) The card will only sync up if we clock, if we line side clock the card goes into yellow alarm and won't sync up. the only errors we see are framing slips. Around 2500 slips over a 18 hour period. (this was reported from

Details: Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: India This is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the zaptel.conf, trying zaptel v1.2(with

Hi, I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is: [from-sip] exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN}) and my zaptel.conf is: fxsks=1 loadzone=us defaultzone=us and my zapata.conf is: context=incoming signalling=fxs_ks echocancel=yes