similar to: zaptel: [rx|tx]gain on E1/PRI/isdn audio quality problems

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud).

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

I can't build it, errors: chan_zap.c:61: #error "You need newer libpri" chan_zap.c: In function `zt_call': chan_zap.c:1806: warning: implicit declaration of function `pri_sr_set_redirecting' chan_zap.c: In function `pri_dchannel': chan_zap.c:7776: structure has no member named `redirectingreason' chan_zap.c:7778: structure has no member named `redirectingreason'

Hello Robert, ----- Nachricht von Robert Moskowitz via dovecot <dovecot at dovecot.org> --------- Datum: Sat, 9 Feb 2019 22:50:24 -0500 Von: Robert Moskowitz via dovecot <dovecot at dovecot.org> Antwort an: Robert Moskowitz <rgm at htt-consult.com>, Dovecot Mailing List <dovecot at dovecot.org> Betreff: How to backup maildir An: Dovecot

Dear Help,I am running Samba 3.0.25 on AIX 5.3 (installed from the binaries available on samba.org including the base install -- openldap, etc.) and have set it up to authenticate to LDAP directories on two different servers (one of them set up as a samba PDC and the other as a samba BDC) in the usual way:[global]workgroup = mydomaindomain master = no...passdb backend =

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

Reduce special casing of xen_cpuid() by using cpu capabilities instead of faked cpuid nodes. This cleanup enables us remove the hypervisor specific set_cpu_features callback as the same effect can be reached via setup_[clear|force]_cpu_cap(). Removing the rest faked nodes from xen_cpuid() requires some more work as the remaining cases (mwait leafs and extended topology info) have to be handled

Reduce special casing of xen_cpuid() by using cpu capabilities instead of faked cpuid nodes. This cleanup enables us remove the hypervisor specific set_cpu_features callback as the same effect can be reached via setup_[clear|force]_cpu_cap(). Removing the rest faked nodes from xen_cpuid() requires some more work as the remaining cases (mwait leafs and extended topology info) have to be handled

I have put a box together using 2 X100P and 2 TDM400 4port cards. Using the simple setup that Martin posted a few days ago, I have asterisk almost up and running. /etc/zaptel.conf fxsks=1-2 fxoks=3-10 loadzone=nz defaultzone=nz (I have added NZ tone information to zaptel and Asterisk - I'll submit a patch soon). /etc/asterisk/zapata.conf [channels] context=incoming signalling=fxs_ks

Reduce special casing of xen_cpuid() by using cpu capabilities instead of faked cpuid nodes. This cleanup enables us remove the hypervisor specific set_cpu_features callback as the same effect can be reached via setup_[clear|force]_cpu_cap(). Removing the rest faked nodes from xen_cpuid() requires some more work as the remaining cases (mwait leafs and extended topology info) have to be handled

Reduce special casing of xen_cpuid() by using cpu capabilities instead of faked cpuid nodes. This cleanup enables us remove the hypervisor specific set_cpu_features callback as the same effect can be reached via setup_[clear|force]_cpu_cap(). Removing the rest faked nodes from xen_cpuid() requires some more work as the remaining cases (mwait leafs and extended topology info) have to be handled

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1: