similar to: DTMF doesn't seem to get through incoming ZAP channels

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hi, My Asterisk CVS is apparently not doing much (other than keeping SIP & IAX2 registrations alive and doing some ZAP calls (without echo-cancellation), but slowly the memory is filling up, so much so that 100m virtual memory is used up within 12 hours and I have to restart the asterisk application every 48 hours to make sure I have enough memory... How can I help resolve this problem?

Hi, I am trying to post this again as I am getting no answers and the support@digium.com bounces... (I have searched the whole list and can't find the answer either) I have installed a 5 user license for G.729 and want to route calls through Asterisk from my G.729 phone to Cisco AS5300 also using G729. Both Cisco and the phone connect using this codec if I do not force the call to go

Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything "seems" to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

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

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

I am having some troubles with DTMF detection on zap channels when the remote caller is calling from a noisy cell phone. It is actually detecting multiple DTMF tones (usually 2 or 3) when only one is sent (i.e. I press '3' and Asterisk is detecting that as '333'.) I don't know the exact situation with regards to signal strength, etc, but it seems to happen more often from users

Hi, I'm getting unusable DTMF detection with DISA on incoming ZAP channel (bristuffed *) on quadbri from GSM gateway. DTMF detection works ok in normal ISDN incoming line. How can I check what's going on ? What settings to check ? Anyone with more experience on such scenarios ? Thanks in advance, regards, Rob.

Hi. I have the following line in the default context of all my internal extensions: exten => 9876,1,Transfer(125) When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want. Unfortunately when I dial 9786 from my Zap connected analogue phone, the transfer doesn't go through and the dialplan drops through to a hangup. debug

We're still having problems with DTMF detection on our X100P cards. Incoming callers that hold down the "1" button for too long are being connected to extension 11. One would think fat fingers were uncommon, but it happens to alot of people. I suspected this was related to our having to increase the txgain, but I tried turning it down with no effect. I also tried disabling the

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Hi, can anybody tell me how the Hausman test for endogenty works? I have a simulated model with three correlated predictors (X1-X3). I also have an instrument W for X1 Now I want to test for endogeneity of X1 (i.e., when I omit X2 and X3 from the equation). My current approach: library(systemfit) fit2sls <- systemfit(Y~X1,data=data,method="2SLS",inst=~W) fitOLS <-

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. On Thu, 29 Jul 2004, Walter Klomp wrote: > However, if I dial-in from the SIP phone to my

On 1/9/22 15:37, Gordon Messmer wrote: > 1: The system also includes a volume group named "BackupGroup" and > that group activates on boot (post-dracut).? Why are those LVs > activated when rd.lvm.lv is specified? As far as I can tell, this is because in the dracut boot process, the device backing VolGroup is activated, but the device backing BackupGroup is not.? As a

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've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Hi, I have successfully compiled spandsp 0.0.2pre10 with * 1.05 which can accept inbound fax calls. However, all fax received are incomplete (the first 10% of an A4 page is fine, the remaining is either missing or garbled). I suspect this is due to 'training error' (see below) which, according to Steve Underwood's postings, cannot be resolved further. I wonder if it would help to