similar to: Asterisk integration with Meridian 1 Option 11 / ISDN30

<p>I am new to the group but have searched the doc's FAQ's etc before posting here.</p><p>&nbsp;We are attempting tie our asterisk server/service to the building's PBX, the building is in the UK and the local PBX is a meridian option 11 installed and mainteined by BT.</p><p>BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter cards

Hi all If I have the following in my dialplan: exten=>25001,hint,SIP/25001 Doing a core show hint 25001 results in 25001 at local : SIP/25001 State:Idle Watchers 0 1 hint matching extension 25001 in the Asterisk CLI. What does the Watchers 0 mean? I use the hints table output via core show hints for logic in my dialler application - but

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

Hi, Any of you have any experience with SIP softphone dialler that capable of local recording? (recording to files in harddrive) So far I only know eyeBeam and Express talk. eyebeam fine but there are known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *. Cheers Benni-

For some reason we are seeing "Avoiding deadlock for channel" in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are "Avoiding deadlock". What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's.

Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is

We are having a situation on our dialler here where our agents are claiming that when they receive a call because it has been answered, it seems as if the call had been answered several seconds earlier - IOW, they are hearing "hello ? Hello ?" and often hear the phone being put down as an initial part of the call. We have verified this by checking the voice recordings. Yet, the logs of

Hi, The PBX recives alarms from the TE110p card and are mainly pointing at frame errors and Loss of signal. Asterisk is configured as Zapata.conf signalling=pri_cpe switchtype=national rxwink=250 channel => 1-15,17-31 Zaptel.conf .... This is what I need to know - the SPAN is currently set to E3 - does anyone know what I need to use for a E1 ? span=1,1,0,esf,b8zs bchan=1-15,17-31 # set this

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

hi .. i require to interface asterisk to a 60 line analog PBX in a hotel. I was thinking of giving Asterisk a couple of PBX lines interfaced through cards, and then place outgoing calls through SIP/H323 and a DSL connection. analog extension lines <--> analog pbx <-->asterisk <--> SIP --> termination I do not need incoming calls to the lines. My question is this : if I

I have a samba server acting as a domain controller. Is there a way that I can Have a script that delete the shortcuts on the desktop,quicklaunch and startmenu for Internet Exploder. At the same time installing Mozilla Fire Fox. Maybe like a little vbscript or something that gets ran from the server when they login. Thanks

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

I was playing around with the Flash Operator Panel, and came smack into a brick wall. We have a * box linked to a legacy Meridian System using a EuroIDSN link (TE405p) with 10 channels enabled. I also have several SIP extensions. What I wanted to do was to have a button for each of our (say) 32 users, 5 of which are on SIP. That leaves the other 27 on Zap. A potential of 27 users on 10

On Sat, Jan 7, 2017 at 2:30 PM, Peter Moody <mindrot at hda3.com> wrote: > so I spent a bit of time looking at this and it seems like the only > way to go, at least if I want to keep it in ssh_connect_direct(), is > to use pthreads. further, it seems like getting that accepted is > something of a long shot: Sorry, pthreads is a non-starter. I would have thought that using

I am seeing a BUNCH of these type of messages in my log.nmb file. the IP address varies, with only 2 different ones showing up. process_node_status_request: status request for name *<00> from IP 206.139.7.109 on subnet REMOTE_BROADCAST_SUBNET - name not found. The other IP address is 192.68.22.214 Unfortunately, the log does not have time stamps, so I don't know when this is

Hello Sir, I have few clarifications as we are planning to work with asterisk. If you don't mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and it's usage). Q2. From the list of features listed in www.asterisk.org , I see "Predictive dialler" is listed

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the