search for: sylantro

I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and o...

...tworking component as well. On a side note, apparently my current employer tried a Cisco solution before I came along, and I hear all the time how absolute crap is was. Is that how people who have used Asterisk feel about Cisco? Is Cisco that bad? Is it lacking in features? I know we investigated a Sylantro solution and I remember that was pretty nasty. Anyway, based on the absolute dominance of Cisco it almost seems like what I have been doing with Asterisk has been a complete waste of time from a career perspective. I'm not sure how I can use Asterisk to my advantage over Cisco here. Having mo...

I am located in the UK, and I am using Sipura spa-2000 adapters to connect analog phones to a voip network. The network connects to the PSTN as well via the Sipura spa-3000 adapter. I would like to provide surge protection for the spa-2000 and the spa-3000 adapters. 1. For spa-2000, fxs port: What is the maximum tip-to-ring voltage before damage to the the adapter occurs? 2. For spa-2000,

...eferrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: "Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP Platform partners and at this time those Partners are Sphere, Broadsoft, Sylantro, and Interactive Intelligence. Unfortunately we are not supporting the Asterisk solution at this time." I am going to continue to pursue this, but this is the pushback I have gotten thusfar. We are a Polycom authorized reseller, and have compiled some pretty detailed documentation of worka...

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

So are there any plans for bridged line appearance support in Asterisk? The new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel, Sylantro etc. supposedly support it. Seems to me that Asterisk needs to get on the bandwagon or be relegated to call centers, specialized voicemail applications, and phone chat businesses. It's not needed for companies used to PBX's but something like 75-95% of all companies are small businesses...

Hi all, We are in the process of selection IP Phones to work with our *new* Asterisk PBX. We want to buy 4 for something less than 1000$ but with a nice set of features to work with our mail box, lines, good sound quality, full duplex (and maybe speaker phone). Any suggestions for something with good voice quality and not much troubles to setup with Asterisk? Voici quality is the most

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I