similar to: My first E1 card is running :)

I've reading this lists few months. We are small company, that makes some system intregration, development and deployments in VoIP scene. Completely under linux. Today i have 6 machines with asterisk, huge test base - including devices like AS5350, Audiocodes gateways, ATAs, IP phones ... Now is time to make a decition for including * in our future projects. Main goal for us is the Stability.

Gabe Bourque wrote: > Hello Anton Tinchev, > > I'm writing to you in hopes you can answer a few questions regarding > Asterisk/Digium and it's reliability. I saw your posting in the > Asterisk mailing list (Re: [Asterisk-Users] Is Asterisk ready for "real" > use?) and decided to write directly to you. The reason being that you > are one of only a few

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

The hardware we are planning to use is: Micronet SP5050 FXO Gateway http://www.micronet.com.tw/Products/VoIP/SP5050.asp Micronet SP5100 IP Phone http://www.micronet.com.tw/Products/VoIP/SP5100.asp We are hoping to use this hardware along with AsteriskPBX to replace our aging PBX system. What I want to acheive is: * Any incoming call from PSTN (via gateway) rings on the receptionists phone for

Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a "sip show peers" it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

I tried setting up a Micronet 4port 2 fxs 2fxo device using sip. I found that I could get multiple fxs ports to register with *. I also found that no matter what dialed on as an fxs user when dialing out it would try to go out the fxo side of the micronet. I also had major problems with these devices behind nat. I was having so much fun with these little boxes that I put them back on the shelf

Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is Transferring of calls, and the ability to put calls on hold. The phones have both a 'Transfer'

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

I really hope that someone can help me with this one. DTMF tones are not working for certain places that I call, specifically 1-800-882-8880 which is the AA advantage line. It works for almost everyplace else. If I bypass asterisk, the call works fine. Network looks like: <SPA-2000> --SIP-- ASTERISK --SIP-- <AS5350> --PRI-- PSTN sip.conf entries [VGW01] (this is the AS5350)

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless "HDLC Bad FCS" errors. I modified logger.conf to get rid of the messages (so I could see what else was

Hi, Has anyone out there managed to do a hookflash transfer with a Micronet 5050s gateway ? We have just tried out this gateway and it seems to do everything we need except this particular feature. Also if you have succeeded where is the hookflash time specified in the gateway. There does not appear to be any parameter for this. Perhaps it is not supported at all. Any help appreciated.

I'm first to buy 5 pack. Even for > $30.

Hi, Just wondering if anybody has encountered a similar problem as I have with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have dtmf relay configured on the AS, however, when someone calls in from the PSTN sometimes their digits are inputted twice, which messes up the extensions. If there is a better way to terminate calls from a AS without using SIP, that would fix this

Hi, i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco AS5350)and user is connected via sip too. When user calling out (via AS5350) he receives progress tone generated by voip-phone not that passing from telco line. I turned on debug and see that the AS send: 183 Session Progreess but to user is sent Ringing, not progress. I have progressinband=never in sip.conf so

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the