similar to: OT - IP Network Call Recording

Good day all Is there someone that's got asterisk working well with a A101/E1 card Apparently they don't have RBS support? Please advice Thanks Altus

Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews

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 -

Guys. I know there are wifi sip phones out there but I have a question, are any of these phones "anti explosive"? By that I mean, there are certain regulations about phones or cel phones that are not recommended to operate in environments like gas stations due to sparks and the chance of ingiting gas fumes. Are there any wifi sip phones out here that have complaince with regulations to

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.

Hi all, For a top quality setup, I will need to install high quality VoIP switches with QoS and PoE. My potential customer should not have any problem with call quality. Experienced folks, Please advice me what switches to install and at what price. I may need it for upto 100 phones. What else should I consider so that phones work without problem along with the computers on the same network?

Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DID's assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work,

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

I am looking for a good analog desktop phone to use with asterisk and my sipura ATAs. I know I want Caller ID, MWI, a few programmable buttons (for asterisk features), and no external power supply (so my users can dial 911 through the SPA-3000 when the power is out). I spent some time looking at the phones at Fry's today, without finding exactly what I need. Do any of you have any

Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call

Hi Guys, Im planning to purchase Supermicro server but I dont have idea what Motherboard and chipset model will work with TE406P.I read one post here that the chipset should be Intel E7520.Is it true?Is there other recommendation?My requirement by the way is to handle 1000 simultaneous calls. Does dualcore xeon (single cpu) and 2gig memory can handle this w/o

Don't want the opening line of my post to be misconstrued. "Anyone familiar with the BrookTrout TR1034 Fax Board, supports T.38 and PSTN fax in an Asterisk environment?" That's a question, not a statement. Curious if anyone has played with one of these as an extension of an Asterisk server for T.38 faxing, did not mean to imply it could, should or would play well with

Does anyone know if the Sangoma A101 supports DS4 and/or RBS ! Shame on me for not knowing the answer, but I don't, and I can't find any mention of it on Sangoma's website. Cory Andrews Purchasing Manager ++++++++++++++++++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 ++++++++++++++++++++ direct - 716.250.3402 mobile - 716.907.4054 email -

Has anyone had any experience using FXO and FXS gateways to extend legacy PBX extensions to remote users? I have a customer who needs to do this, but wants seamless, two way communication, with a SIP server and without the need for 2-stage dialing. If anyone has any experience with a solution please let me know. Thanks Cory Andrews

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

Hi, I have problem with runuser utility and permission to /dev/null. runuser is called from init script. When init script is run /etc/init.d/callrec start, then there are no problems with permissions, but when init script is called via service callrec start. Then runuser starts complaining about permissions. runuser: /dev/null: Permission denied What is strange is that I do not redirect to

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin.