similar to: FXS Help Needed...

Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/f6d5af5a/attachment.htm

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

> Hello All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

Hello; If I have load up to 220 extensions with 50 concurrent calls. Can one hardware server carry all this load? What is the hardware server required for this? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/b63756d2/attachment.htm>

Hello Everyone, I have an installation where the client has a Microsoft SQL database that holds all of their case information. They would like the asterisk system to require users to enter a valid case number when making an outgoing call. I?m seeing some documentation regarding people using Microsoft SQL for CDR storage, however nothing regarding validating authentication using a Microsoft SQL

Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core

Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071217/ae6cba5a/attachment.htm

Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Greg, it's easy: Configure searches for the original C++-Compiler, "cc". I presume you're using gcc (I do so on SUN Solaris), and it should be no problem to make configure see what it wants: Just link "gcc" to "cc" in /usr/local/bin (or whereever "gcc" is installed), and it should work, 'cause the two compilers are fully compatible. BtW: If

Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==>Zap * ==>FXS * ==>Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure

Hi, I am building for a customer an * solution that will use 2 Digium cards. 1 x TE110P (T1 ISDN PRI) 1 x TDM40B (4 analog ports, 2 for faxes, 2 for extensions) The system will be connected to the PSTN through the T1 ISDN PRI interface. All customer extensions will be SIP phones except 2 fax machines that will use the analog FXS ports on the TDM40B card. I have been investigating on this

Does anyone know of a good VOIP dialtone provider in the northern Chicago area. My client has tried Broadvoice and Mix and is having problems with latency in the middle of the traceroute between him and the provider. Thanks, Greg No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date: 1/4/2008 12:05 PM

Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built

Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a "squeaky static" -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this

Jon Pounder wrote: > > <snip> > Fax is really the only need recently, and even that has alternatives like emailing scans that most people prefer now. > The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is fearful of the legal beagles. These groups are really slow to change. At least in

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Hi there, I'm trying to get hold of an evaluation IP-enabled FXS channel bank. I'm not sure if it's more a channel bank, or should be called a multiport-ATA. Oh well. On the surface it looks quite nice - 16 FXS ports, and since it's connected to the network I can do away with an E1/T1 connection required of the normal channel banks (if it can be called that :) Here are some

Hello all, Normally I just use pri's with our asterisk systems, but a request came in to add some normal pots lines to the setup. We have 3 lines, and they run into the fxs ports. They hit the dialplan just fine, and they always dial the "s" extension. However, my question would be... Is there a way to determine what number was dialed and have it forward to a specific phone? With a

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.