similar to: Astersik vs. Pingtel

1) Who bought Pingtel's phone line? 2) Anyone seen this chinese-made VoIP phone that supports 8 different protocols? http://www.telecom.globalsources.com/GeneralManager?language=en&design=c lean&action=GetArticle&article_id=9000000055338&page=printarticle&printT his=yes Mike :) -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Has anyone compared * to sipXpbx? From a cursory look, this open source version of PingTel's PBX has many features that make it more suitable as a replacement for a traditional PBX, including the ability for users to tell if a phone/trunk is in use. What I am trying to figure out is what I'd give up using sipX instead of * (and vice versa). /carmi

Hello all, Hope I am not too of topic here - but it cross's the phone/asterisk boundary. I have been playing with a few soft phones - noticed that pingtel seemed to be highly recommended across previous postings. I have been using xten - which is a great phone but seems a bit limited in its functionality - which is why I am now looking at pingtel. Problem is I cannot get it to

Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero

I just bought a Pingtel Xpressa from VoipSupply for use with Asterisk. I know that Pingtel has sold off their hardphone line and discontinued support for their phones, but I'd like to track down a few of the Java applications that they distributed before they went away, specifically their LDAP Phonebook app. Does anyone have a copy that they could send me? It was publicly

If you can in any way improve this page: http://voip-info.org/tiki-index.php?page=Pingtel+Hardphone please do. Thanks very much. Jason Sjobeck -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/d9b379e1/attachment.htm

In the past two days, I've been experimenting with the Pingtel SIP softphones and Asterisk, which one of my customers has been using. A few notes: 1) The "insecure=1" setting in SIP peers now works, from my limited experiments. Mark put this in for SIP servers that don't send requests in with a return port of 5060. This flag essentially takes any request inbound _to_ port

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to place calls on hold using Asterisk Manager Actions? Amaury -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030906/ed1d46cf/attachment.htm

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

Dear Sirs, I had compiled PWlib and OpenH323 correctly in my Fedora Core 2. But when I try to compile asterisk-oh323 I get the following error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' How can I solve it? Thank you for your help. Juanjo

Hi, I am new to the Asterisk world. I don't know much about the architecture, but I am involved in installing and configuring the VoIP system. My requirement is to build a VoIP system using the 4 input lines (ISDN up0 telephone lines), it must be possible to receive calls from outside through the 4 ISDN up0 input lines, and also possible for outgoing calls, conferencing .etc. I

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client:

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf ============================== [ext-queues] include => ext-queues-custom exten => 5000,20,Macro(user-callerid,) ; changed the priority to 20 ............... ============================== In extension_custom.conf