similar to: Pingtel applications

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

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,

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

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

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

All: Any have sample config's for an ADIT600 w/ CMG card connected to an Asterisk box via MGCP? Thanks, Garrett Smith <mailto:gsmith@voipsupply.com> gsmith@voipsupply.com VoIPSupply.com -- a division of B2 Technologies, LLC (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales VoIPSupply.com is North America's largest VAR

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

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

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:

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Hi all, [I'm new on this list. On other lists, I cringe at this type of query because they sometimes end up in flame-wars. This is a serious request because my time frame is about 2 months to identify, select, acquire, install and setup our next PBX at a new office. I really would like it to be a VoIP solution.] Our small real estate firm is experiencing explosive growth lately and

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi

I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -

Hello, I wrote a small AJAX phonebook targeted at Grandstream phones although the basic functionality doesn't require a SIP-phone. Asterisk integration (call history, incoming call info, click-to-dial) is not yet implemented, but on my ToDo list. Features * Licenced under the GPL v2 * Uses gettext. So translation should be easy. Includes German and English. * Uses xajax for a

Hello, I updated my AJAX GS/* phonebook. It now actually has Asterisk integration with incoming call-notification, a searchable call history and clickable BLFs. The BLFs can monitor specific extensions (SIP/phone - may work with other channels than SIP if the AMI generates the appropriate events), Asterisk hints or connectivity to external SIP-servers. Features * Clickable BLFs (Dial,

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence and virtually no ability to use DTMF in IVRs. I really wish the TDM22B worked, because I much

Hi all, A new DIAX version 0.9.10f is available for download at one of the followind locations: http://www.laser.com/dante http://www.cosmica.ro/dante http://www.geocities.com/tdanro What's new in 0.9.10f comparing with the old 0.9.10a: - volume and mic levels saved in the config file; - master and wave levels adjusted in the same time for the play device; - better handling registration