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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, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using

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, 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, with asterisk 1.6 i am trying to make a dialplan Which i have such entry in extensions.conf exten => _8XXX,1,Dial(SIP/${EXTEN}) But some of my clients have both IAX and SIP accounts, to use iax clients while outside of my Local Area, and SIP clients (or hardware phones) in local area. But with such rule, i can only dial SIP accounts. Is there a parameter to find how the user connected?

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

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 ;

Starting from today i am receiving the following errors on asterisk.. What can be the reason for it? [May 7 11:45:16] ERROR[14885]: chan_dahdi.c:10515 dahdi_pri_error: ACK received for '0' outside of window of '3' to '4', restarting [May 7 11:45:16] WARNING[14885]: chan_dahdi.c:3347 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel

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, I wanted to add a failover trunk to my asterisk configuration. I got 2 gateways for my calls.. one is a pri other is voip trunk. I want to keep my trunk for failover. I am using ast 1.6 with asterisk-gui. But when i add a failover trunk for test purposes asterisk-gui adds the following line to my extensions.conf. where superonline is my voip provider and span_1 is failover trunk. exten =

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi, I want to activate DND on ast 1.6.0.9 with asterisk-gui. Is there special commands that i need to use during such script or simply writing a code in extensions.conf that checks if the user has a DND=yes value on ast. database and act according to that like forwarding call to voicemail or sending back a busy tone or playing a DND msg. And is there a way to notify a GPX_2000 for example for a

Hi, I suddenly started to receive an error like [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler [Aug 4 14:26:57] ERROR[2477]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? and it went on till i reboot asterisk and dahdi services.. I wonder what caused this error, because there were no physical problem between pbx and asterisk

Hi, I got ast. 1.6.0.10 working for a few weeks without a problem. A few mins ago..I got the following msgs on ast-cli and asterisk service crashed. I coudlnt find anything that might cause this problem. Any ideas?? [Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein: Invalid GSM data (1) [Aug 7 13:54:34] WARNING[10102]: translate.c:204 framein: gsmtolin did not update samples 0

Hi, I am getting the following messages for a few days. WARNING[10148]: chan_iax2.c:1219 __send_lagrq: I was supposed to send a LAGRQ with callno 15691, but no such call exists (and I cannot remove lagid, either). I got 2 iax trunks which used by test purposes only. And no iax clients or no calling rules thru this iax channnels so far.. So why do i keep getting that messages.

Hi, Whenever one of my trunks becomes unreachable or reachable again.. On logs i got the msg as follows: Jul 31 15:15:51] NOTICE[15112] chan_sip.c: Peer 'voiptrunk' is now Reachable. (12ms / 2000ms) [Jul 31 15:15:51] WARNING[15112] res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. I dont wanna turn on the debug function because theres a lot of

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, I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql. Everything seems workging correctly except cdr logs. It fills up all data when a call established except src and clid Wht can cause this and where should i check??