similar to: Asterisk as VoIP gateway

I'm currently looking 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

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Hi guys, Can somebody help me on some questions please ? If I have a VoIP network with my Asterisk platform in Europe, what do I need to interconnect my VoIP network to another network in the USA in order to my customers in Europe be able to call to customers in the USA network ? The network in the USA is not an Asterisk platform. Thanks,

Hi I have a system (S) that has a PSTN gateway to accept incoming calls and setup outgoing calls from/to Telco network. In the other hand I have a distant Asterisk box (A) that I would like to connect to (S) using the PRI interface. I understand that the proper way is to order to my Telco two PRI lines one for (S) and another for (A), and configure (S) and (A) to call each other numbers when

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110614/42cbd90d/attachment.htm>

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What

Hello list members, after programing of dialplan I have some messy Custom:hints which I can see in 'devstate list'. I didn't find any possibility how to remove this hints from Asterisk and I want remove them.? Can you help me with that, please? I tried search about that something in documentation or on Google, but I didn't find anything.? asterisk*CLI> devstate list ?

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users. Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM. SIP-Phones

From: Bruno Delbono <bruno.s.delbono at mail.ac> > I'm really sorry to start this thread again but I found something very > interesting I thought everyone should ^at least^ have a look at: > http://uadmin.blogspot.com/2005/06/4-dual-xeon-vs-e4500.html > This article takes into account a comparision of 4 dual xeon vs. e4500. > The author (not me!) talks about "A

When using OCFS2 in a Oracle RAC set, one would configure OCFS2 to use the private interconnect address ( in cluster.conf). Are there any known issues / reasons not to have the private interconnect set up for jumbo frames when using in conjunction with OCFS2 ? Thanks -- -------------- next part -------------- An HTML attachment was scrubbed... URL:

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