similar to: Problem in making outbound calls in PRI

Hi all, I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of command DIAL when used from the dial plan. Can some one guide me how to get the vaue of

Hi. recently I have been trying to setup a PRI on asterisk. Inbound calls are working just fine but I am not able to make outbound calls. Does anyone know what I need to change to make outbound calls work? Right now the PRI is instantly hanging up on the outbound calls. I have included full debug info as well as config files. /etc/zaptel.conf span=1,1,1,esf,b8zs bchan=1-23 dchan=24

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

All We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. National Numbering Plan for Landline in India is typically 0+Area Code + phone number. If it's a local number, you

Hi all I wonder if anyone can give a little insight into this; asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless, ISDN30. 10 zactive channels Incoming calls work fine no problems tested 43 DIDs all working. zaptel.conf # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 zapata.conf [channels] language=en context=from-pstn

Hi, I have a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>

Hi team, On event of no answer in CDR the starttime and endtime of call remains the same. Is there any way how can actually track call originate time and call end time. Thanks Vinod dharashive. Sent from BlackBerry? on Airtel

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

I've got a strange issue where I get no ringing on outbound PRI calls using the TE410P span 1. The call actually goes out and works, you just hear no ringing. The quality and features on the call are pristine, no cracks pops or any weirdness like that. If I specify the 'r' option on the Dial() string, I get a half of a ring, sometimes a full ring as soon as the dial completes.

I'm doing these manipulations on the data frame and wondering why does R have to remember historical data on my operation and not just keep the needed info. Probably a basic fundamentals of the way R handles data .. Pls point me to the manual if possible .. I have this Index data: > head(NIFTY_INDX) Constituents.list.of.S.P.CNX.Nifty X X.1 X.2 X.3

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hello list We're trying to get a HFC card running in NT mode. Zaphfc loads fine as module and ztcfg -v is run only once per reboot. Cabling is ok, the ISDN-TE gets ist power fine. The HFC card does not share any interupts. After the TE picks up the console (pri debug enabled) shows: Nov 21 18:42:20 NOTICE[13326]: chan_zap.c:7169 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharashive -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hi, I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using asterisk from ubuntu linux. Everything is working as expected. This box is being used as a H323 gateway to the pstn. There are few complains but it is working pretty well overall. There is one thing that is bothering me. Asterisk says: May 22 05:03:39 WARNING[9360]: PRI: !! No channel map, no channel, and no ds1?

We have just had an ISDN PRI service commissioned here in AU (using Powertel as provider). I have called them and ensured that we have the ability to set Caller ID on our service to any number in our 100 number block, and I have been assured that everything is OK from their end (unlikely). Every time I drop a sample.call in to /var/spool/asterisk/outgoing with the Callerid: option set, the