search for: infoway

Hi, We are currently trying to setup Asterisk with iBasis. One question/problem we have is that Ibasis has told us to send the INVITEs to one IP address and all media to a different IP address. How can we do that in Asterisk? Thanks

Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to

We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be

Is there any documentation you guys can point us to in order to learn more about the proper use of the Local channel? We don't currently use it. However, while evaluating other people's billing and management systems for Asterisk, we noticed they make extensive use of it. Thanks, Daniel

Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks

I just upgraded my 7970G to the SIP firmware. What I'd like to do is have the 8 line buttons be able to make outbound calls using the same account (for practical purposes, same caller-ID). Since the phone is going to have a single public DID, when a call comes in, it should ring on the first available line. So, if I'm on line 1 and a call comes in, it should ring on line 2. How can this

Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

Hi, We have a "simple" AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? Why is this affecting the VoIP world so much and not the regular PSTN analog world? What does the PSTN industry have that they can handle such high volume of calls and there is "no" echo problem? Thanks, Daniel

Hi, We developed a somewhat complex Asterisk-based IVR for a client of ours. When I say 'somewhat complex' I mean that this IVR is highly integrated with a set of middle-ware applications that allow the caller to perform extensive functions while navigating the IVR. This has been working great for the past year or so. However, now the client wants to add the ability for callers to be

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Hi all, We want to offer hosted PBX services to some of our clients (maybe 10-20) and were wondering if it makes sense to get a software package capable of handling multiple virtual tenants or if we should just create multiple virtual machines in our server each running a single- tenant license of the software. We have been researching virtual PBX software for asterisk for a couple of

Hello all, We have a slight issue to resolve. We have a client who we are drafting an SLA for the delivery of telephony services using Asterisk. Nothing extraordinary. However, we do need a way to measure our service availability. We currently use Nagios and Cacti to monitor server availability as well as asterisk and mysql responsiveness, and last, "ping" availability to our

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

I've read in different places that if I want to do trunking and meetme on Asterisk I need to have a reliable timer. People have recommended that I install a Digium board, even if I don't have any circuits connected to it, just to get a reliable timer. However, I've also read that if I'm using kernel 2.6, I don't need to have a Digium board. I have a few servers that

I have an Perl AGI script that acts as an IVR for my Asterisk box. Basically, it simply plays audio files to the caller, collecting DTMF input and logging every DTMF input into a database table, simply to document every step or option selected by the caller. One thing is that in addition to playing audio files, it also, at some point, plays SayUnitTime command. This IVR has constantly

I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the

Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel