similar to: asterisk sip with voicemail

Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so

Yes, I have the exact same problem. :( -----Original Message----- From: Tomislav Vojvodic [mailto:tomislav@vox-mundi.net] Sent: Thursday, May 11, 2006 5:48 AM To: xytek@hotmail.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted

Hello all, I am a newbie to this list - and so far very impressed with the functionality of Asterisk. So far I have setup a simple soft phone running on a windows PC making calls to other SIP soft phones. Later this week I hope to get UK ISDN2e up and running with it! My question is I would like the experience and feedback from users about what equipment/software you are all using for

Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick

I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to

First I like to apologize if this is common knowledge, but I'm unable to get MSN messenger 4.6 to register with asterisk. I configured MSN messenger to use UDP and the IP of my asterisk server I edited the registry entry - for pC2PC calls under Windows98. What I'm I missing ? Asterisk version information Asterisk CVS-04/25/03-05:37:19 sip.conf [pingtel] type=friend

I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? Another question, when I recive a phone call, can that

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Dear User, I got the following error running a regression quantile: > rq1<-rq(dep ~ ., model=TRUE, data=exo, tau=0.5 ); > summary(rq1) Erro em rq.fit.fnb(x, y, tau = tau + h) : Error info = 75 in stepy: singular design Any hint about the problem? Thanks a lot, ________________________________________ Ricardo Gon?alves Silva, M. Sc. Apoio aos Processos de Modelagem Matem?tica

Dear R-Users, How can I compute simple moving averages from a time series in R? Note that I do not want to estimate a MA model, just compute the MA's given a lenght (as excel does). Thanks ________________________________________ Ricardo Gonçalves Silva, M. Sc. Apoio aos Processos de Modelagem Matemática Econometria & Inadimplência Serasa S.A. (11) - 6847-8889 ricardosilva@serasa.com.br

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/59c113df/attachment.htm -------------- next part -------------- Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card,

Hello all, Does anyone has experience of deploying Asterisk based VoIP solution in a universitywide campus. We are at present investigating various Soft PBX for this purpose from different vendors Digium,Snom, Pingtel... We are looking at serving more than 5000 clients and we want to be very sure before taking any final decision. I would be glad to hear from members who are aware of

Hi, I load my data set and separate it as folowing: presu <- read.table("C:/_Ricardo/Paty/qtdata_f.txt", header=TRUE, sep="\t", na.strings="NA", dec=".", strip.white=TRUE) dep<-presu[,3]; exo<-presu[,4:92]; Now, I want to use it using the wls and quantreg packages. How I change the data classes for mle and rq objects? Thanks a lot,

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back

Hi, To make a long story short, I can't install any TAPI driver on my XP platform. A. Within Config Panel|Modems and Telephony options|Advanced parameters, I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for Asterisk. B. I can properly configure this driver (line, context, ...). C. When I open Outlook 2002 Contacts panel, I can select "Call this contact"

Hello! I have finally ordered some Asterisk hardware: the TDM DevKit. However, I want to use VoIP phones (or possibly adapters) for remote users. I would like to get some suggestions on which phones to buy. I'm hoping that some of you with real experience might be able to help me out! Here are the features that are important to me: * While these phones are initially going to be

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist