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Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to

Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so

Hi, I am using asterisk 1.6.0.10 For debugging i set verbosity to 10 with asterisk -vvvvvvvvvvr.. now i am trying to set it lower but.. when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10 currently running on asterisk1 (pid = 2408) Verbosity is at least 10 when i try set verobisty 1 or similar commands.. i think this command is obselete in 1.6 .. set verbose 1 No such command

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson

Hello, We have a sip trunk between our voip operator and our asterisk 1.6.2.9 We have no problem during voice communications. But we can not send any t38 fax via this gateway. We tried to trace the error made some tests.. There are 2 main tests we tried to do. As i learned their voip path is like .. we connect to session border controller..then it routes the call to a cisco media gateway if the

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hello, There is a problem that i can not figure out how to solve. I got users with 5 digit usernames for sip. Some users has a callerid for outside calls. I have such problems When a user activates (for ex) call forwarding, System creates that entry on database as CFIM/callerid not the username, So this rule works only if a call is made from outside to the callerid. Not the local calls made

Hello, I was testing my asterisk for a while with 1.6 without much problem. Now i am trying to install a new system with asterisk 1.4 but now i am using a dual pri card instead of single pri.(TE220P) What i want is to use both PRI ports as group. Now i have zaptel.conf file created as follows -------------------------zaptel.conf-------------------------- # Span 1: TE2/0/1 "T2XXP (PCI)

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1111 1112 1113. What i want to know

Hello, In our previous PBX we have an option to turn off or on outside calls with a pincode.. Like, user is able to get calls or dial local lines by default, but when he/she uses a password entrance via dtmf, he can dial long distance calls etc.And at anytime he can logoff from outside call permit.. So is it possible to do smthing like this on asterisk.. A limited profile which needs sip password

Hi, I have the following dialplan. Problem is, if the user authenticates, * starts counting as billable seconds even if i hangup the phone before the called party answers..And also as disposition.. it accepts all calls authenticated as 'answered' If i commentout the authentication line everything works as it should be. How should i use authentication that, it should accept it as aswered by

Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can