search for: dialler

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-

...| | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri) | | | Melita predictive Dialler | NTAK79BC (2MB Pri)|<===> +---------------------------+ | | | | | NTAK79BC (2MB Pri)|<===> | Aculab E1 SC-BUS ISA | | | | (Dialogic D320/SC) |...

.../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 what is a "watcher" in the context of dialplan hints? Is this something I can make use of? My dialler app uses SSH to get the asterisk -rx output of core show hints and then string processes the result to determine which extensions are busy, et al. Can I for example so...

...-- I think you're overcomplicating your problem. (if I understand you correctly!) Your scenario is almost exactly ours, except we use ATCOM-820P's (with LCD displays) and no softphones. So incoming CID is displayed on the phone's physical LCD displays. What we did is write our own C# dialler app - all this effectively does (through a third-party server app we designed) is connect over the AMI to the Asterisk instance and then use the "originate" function to originate a call to the user's phone. Behind this is a database where we store which logged in user in the dialler...

...e 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. [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c:...

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

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

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

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 verif...

Hi, I have recently started a job as a system administrator, and as part of my responsibilities I have to look after an asterisk system. Quite impressed with it, but have one or two niggling issues. One of the last things my predecessor here did, was install a VoiceBlue mobile gateway unit, and though it seems to work ok, nearl 20 seconds pass from dialling a number to the call connecting, which

...erisk. 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 under "Telephony Services" but Not ACD functionality under "Call Features" Q3. If ACD is not supported, how come "Predictive Dialer" is going to work? Please do clarify me. Thanks & Best Regards, Ajay Kanth Ph: 9848880309...

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira

hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

if the remote hostname has multiple ip addresses, ssh_connect_direct will currently loop and try each address in sequence until one works. I'm interested in making ssh tries each address concurrently and return success on the first one that connects. in the land of host certs and ssh bastions, this can be incredibly effective. are there any objects to me working up a patch to implement this?

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

...ndstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays in the conference room working offhook. 2. Dialler originates calls from the meetme conference room to the target party. Problem... 1. every now and then the agent experiences large amounts of echoing of their own voice (not the target party). It's not consistent, just every now and again. The echo is of quite a long delay... My assumptions....

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

...) would be feasible, though. > so, approaching this from a different angle, what if I wanted to have > something else establish the tcp connection and then fork/dup2/exec > ssh and pass off the fd's for the network connection? That's how ProxyComand and ProxyUseFdpass work. Your dialler is a separate program so it can do whatever you like, including use pthreads if that's your thing. -- Darren Tucker (dtucker at zip.com.au) GPG key 11EAA6FA / A86E 3E07 5B19 5880 E860 37F4 9357 ECEF 11EA A6FA (new) Good judgement comes with experience. Unfortunately, the experience usual...

I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously