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I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this

...= ou=people ldap group suffix = ou=Group ldap machine suffix = ou=people #ldap filter = ($(uid=%u)(objectclass=sambaSAMAccount)) ldap idmap suffix = ou=Idmap idmap backend = ldap:ldaps://host.domain.com add machine script = /var/lib/samba/sbin/smbldap-useradd.pl -w '%u' thanks -- Simone Cittadini ================== COMVERT S.R.L. via F.lli Bressan, 21 20126 Milano - ITALY Tel +39.02.27006796(aspetta un beep)103 simonec@comvert.com http://www.comvert.com

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

let's suppose I have this dialplan : exten => _X.,1,Playtones(ring) exten => _X.,2,Dial(CAPI/contr1/${EXTEN},,g) exten => _X.,3,AGI(update) where "update" updates some db tables we have based on the type of extension Now, from the wiki : If the /g/ option is specified, and the called party hangs up before the calling party, then Dial exits with a return code of 0 to

While I wait for the call to be answered I hear a "double ringing tone", like : expected tone : tuuu tuuu tuuu tuuu what I hear : tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu the second "tuuu" I think is generated somewhere and not "true", since it sounds slightly different and the lambda between the first and the second is always

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine "tuuu tuuu instead of tuuu" we've solved the problem changing the call progress tone of sip phones to something not udible.

I have strange peaks of machine load on my asterisk servers, looking at top the load is very high even if cpu usage is low and no swap memory is used. This happens on all the machines, some of them have asterisk, mysql, agi and digium cards on them, so I thought I was only asking too much, but yesterday I noticed the same behaviour on an asterisk machine with only two digium in it, no other

I've setup this simple configuration to test the new mediaonly iax feature in 1.4 : Input (client) -> Server (routing) -> Termination transfer=no transfer=mediaonly transfer=no all the machines are in the same 192.168.0.x net the routing Server in the middle has iaxusers realtime backend on mysql the call is originated with a sip phone registered on the Input client

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

Can anybody, please, explain me how many parameter are estimated using randomLCA? For examples, model "dentistry.lca2random" estimate 1 scale (or variance, b_j) parameter and 2 position parameters (a_cj)? Doesn't it? Do I need at least 4 diagnostic tests for such a model? What happens if I specify options blocksize and byclass? How many diagnostic tests (or rater) I need?

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds >Asterisk Event Logger restarted >Rotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this

Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models?

I'm trying to install a TE410P this is what happens with compiled zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/ this is my zaptel.conf (checked with the provider the values): span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=it defaultzone=it then I modprobe wct4xxp debug=1 t1e1override=15 and the kernel says : Sep 30 16:12:40 localhost kernel: Zapata

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML