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I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

Hello, I'm having a problem with LMTP in a proxy server since I upgraded from 2.1.16 to 2.2.28. In my logs I have: Apr 26 12:54:17 musio12 dovecot: lmtp(2082): Fatal: master: service(lmtp): child 2082 killed with signal 11 (core dumped). As far as I could check the message is delivered in a later connection. I have enabled core dumped files, but how could I debug it? -- Angel L.

Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -------------- next part -------------- An HTML

I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001

I am having a problem trying to get a Sangoma A101 to work with Unicall. I have installed the sangoma drivers and everything seems to be well but when I try to run ztcfg I get the following error: CAS signalling on span 1 conflicts with HDLC with FCS check on channel 16. Here is my /etc/zaptel.conf # MFC/R2 normalmente no usa CRC4 span=1,0,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101

Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

Hi, I know that there are some variables (as user or username) I could use inside dovecot. They are at https://wiki2.dovecot.org/Variables. My question is if I can create my own variables from attributes at my pass/user db and then use it inside dovecot configuration. -- Angel L. Mateo Mart?nez Secci?n de Telem?tica ?rea de Tecnolog?as de la Informaci?n y las Comunicaciones Aplicadas

Hi, there. I am new at list and I would like to introduce myself: I am from Tuxtla Guti?rrez, Chiapas, M?xico, and I am in Spain right now, studying at university, i am majoring in new tecnology in media. Actually i am trying to mount a radio statio with this software, Ice cast, but I don't understand how to make my xmms player get mounted in live_xmms in ice cast. In fact, i only can

Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not

Hi, I am turning crazy. I try to integrate Asterisk 11.5.1 into Samba4 LDAP, but when I import the ldif file from contrib directory I get this error. ldbmodify -H /usr/local/samba/private/sam.ldb asterisk.ldif --option="dsdb:schema update allowed"=true ERR: (No such object) "objectclass: Cannot add cn=asterisk,cn=schema,cn=config, parent does not exist!" on DN

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

Hi, If I use --enable-fixed-point-debug I can't use --enable-fixed-point and vice versa? Because when I try to copile with these two option an error occur as follows: gcc -DHAVE_CONFIG_H -I. -I. -I.. -I../include -I../include -I.. -I/home/liselene/projects/speex/speex-1.1.11.1/bin-fixed/include -g -O2 -MT modes.lo -MD -MP -MF .deps/modes.Tpo -c modes.c -fPIC -DPIC -o .libs/modes.o

Hi everybody any one knows, how i could edit dovecot to assign user quotas ? I have now configured my dovecot.conf on this way: protocol imap { listen = *:143 mail_plugins = quota imap_quota } protocol pop3 { listen = *:110 mail_plugins = quota } plugin { quota = fs:INBOX:mount=/

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub