similar to: asterisk 1.6.1.0 and dial plan changes

Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

Hi list, I have successfully authenticated active directory users with samba. Now I need to create some Active directory security groups and authenticate and redirect those users to a specific directory. Ex: IT_GROUP - user x , user y FIN_group - user a, user b If the user x , access the samba server, that user will be redirected to the specific directory (that's in the samba stanza). This

Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

hi folks, I use Dial(Local/extension@context) to make calls received on my DID number to ring a local extension. I notice that on 8 out of 10 calls, the audio is NOT working in the incoming direction (DID provider to asterisk). Local extension 2055 maps to SIP destination "homephone", and if i replace the Dial(Local/2055@local) with Dial(SIP/homephone), it works fine 100% of the

I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the following results (same dialplan, config etc): Asterisk 1.6.0.1 => works fine Asterisk 1.6.0.9 => can't dial out unless I dial in once or apply patch >>>==> http://bugs.digium.com/print_bug_page.php?bug_id=14577 Asterisk 1.6.1.0 => can't dial out, regardless of patch or inbound call first.

I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then I go to extension h and have

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hi guys, I have installed samba with AD authentication. Ntlm_auth is working without any issue with the domain. But if I connect using my windows pc, to the samba share, it gives following error. Wbinfo -u / wbinfo -g giving the correct output. And ntlm_auth also working without any issue. If I try to connect from my windows PC to the samba share it gives following error. [2010/12/01

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

Hi, I'm having a problem calling international numbers with debian's Asterisk 1.0.7 w/ zaptel 1.0.9 in Moscow. Russia doesn't seem to have touchtone dialing, so pulsedial is enabled on my TDM400P interface. Local numbers work fine, but when it comes to long distance or international, I'm lost. The prefix for these should be 8 (wait for dialtone) 10 (country code) (city code)

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the "soft hangup Zap/<zapchannel>" in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes

Hi list First of all could you please explain loop-start and ground-start for me? What are they used for? Next, I have the following configurations: dahdi-channels.conf : context=pstn-channels signalling=fxs_ks channel=>130 context=phone-channels signalling=fxo_ks channel=>127 chan_dahdi.conf : [channels] cidsignalling=dtmf cidstart=dtmf signalling=fxo_ls pulsedial=no

Hi, i have two asterisk@hotme 2.2 server. i want if one of my asterisk server down. other is taken control of my first server and call goes through. Is it possible in asterisk. Usman -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051216/7b0c169d/attachment.htm