search for: vegastream

We have a Vegastream 400 connected to a digium Quad PRI card in an asterisk server, for the T.38 faxing here. Problem is that there are too many clockslips on it (and they get logged by asterisk as HDLC aborts). I've double checked the configuration on both sides, replaced the cable, tried different ports etc. It...

I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been...

I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSet...

> > I am looking for "how to" information / references on use of either Audiocodes MP104 or 108, or Vega 50 Gateways for interconecting Asterisk to the PSTN via FX0 interfaces. > > Any info of references / personnal experiences would be appreciated > > Stratus? THE WORLD'S MOST RELIABLE SERVERS * > Richard C. Sparacino > Telecom Technology Manager

...system. Most FXS gateways don't seem to support this... I have a stack of Sipura 2002's that are, AFAIK, worthless for this purpose. I've heard that Digium's IAXy FXS will work with modems, but I can't find any reference to that in their documentation. There is also the VegaStream Vega 50 that claims to support v.90 on 8 FXS ports per unit. Does anyone have experience with these devices, or can recommend anything else? - .Dustin Wenz

...king to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are all expensive (over 3,000). Does any one have any other suggestions or experience with the above products? Thanks, Jameson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipe...

...bound calls, however the issue we have is that when calls are sent to the Sipura with Dial(SIP/${EXTEN:0}@sipura1) the Sipura does a SIP answer immediately and then proceeds with the call "in band" therefore sending dialing sounds back to the caller. Other SIP gateways we have notably the Vegastream and others do not do a SIP answer until the call is successfully connected to the called party. Any ideas?

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

We recently had a customer install that went horribly wrong. Serious echo (pots lines into a cac cb) that, although * did a good job getting rid of alot of it, could not get rid of it all. We tried everything, every canceller, gain setting, etc... combination possible to no avail. Both the vegastream and mediatrix boxes also could not get rid of all of the echo. So, on an off chance, we bought an 8 port fxs/fxo/e&m gateway made by multitech. The echo cancellation on this device is amazing. There is no trace of the echo and the conversation is still full duplex. And, the box works perfec...

Has anyone here using VegaStream FXO with asterisk? I just got the Vega 50 10 FXO and all I could manage by now is to get outgoing calls. Any pointers and a script sample would be appreciated. Thanks, Issac

...h 2 external S0 -> first SO interfacing PSTN -> second S0 interfaching * With this setup what can I use to interface *? 2.) A little box thats an FXO and FXS at the same time: ISDN-phonsystem -> FXO/FXS-box -> PSTN -> * The Vega 50 on www.vegastream.com seams to be able to do this but it is way to expensive for what I'm trying to achieve (saw prices round $2000). 3.) Hopefully a better/cheaper solution one of you can come up with. Maybe using standard ISDN-Cards like AVM B1... I've googled the web for quite a wile to find a solution...

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?

...s like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frust...

...the correct way to route some calls from Asterisk to another h323 > > gateway? > > > > Thank you, > > Mark > > > > > Michael. > Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream ------------ exten => _01XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) ; Crilium --------- exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) Shimul > > > > --__--__-- > > ________________...

...new to all this so I don't know where to look, some tips would be most appreciated. I've enabled sip debugging and everything looks fine on the client and server side. Using Linphone on the client side. GSM playback from the server console is fine. I've used Linphone to connect to a vegastream VoIP system so I know if that installed and working. I'm basically just trying to get the sample configs working, dialing in to 500@ipaddress. It connects and playback of the demo ensues, but at the client end it's unrecognisable garbage. Any hints ? -- Glen Gray <glen@lincor.com>...

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digi...

Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78