similar to: Sip to PSTN Gateway Configs

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

I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the "FXO"-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my private Asterisk-server. Could someone advice on a gateway that can take analogue calls

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

hi all, Im trying to configure a * server with FXO and FXS cards. Basically what i want to do is be able to recieve calls from PSTN and dailout as well...but im really very confused with how to handle the 30 channels coming in on the PRI....what would be the best hardware to use n stuff.. if someone can give me some hints or config examples that would be just great.... Thank's in

hi guys, i am getting today my dev kit with fxo and fxs boards. i intend to do the following : 1) be able to route an incoming call from the pstn fxo port to an ip (answering with netmeeting or anyother sip softphone) 2) be able to call from netmeeting to my pstn fxo port to place calls. questions : how can i do this ? what are the commands for this simple setup ? how can i place calls

I just installed a new asterisk box with a wctdm with 4 FXO modules. The lines in the office have terrible static (using standard analog phones) and this static can obviously be heard through the asterisk box on the sipura sip phones we installed. This by itself would not be a problem as the office is used to and doesn't mind (I don't know how) the static. However it appears that this

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

>From your experience, would you recommend purchasing 8 Sipura 3000 1 port FXO gateways or 1 Audiocodes 8 port FXO gateway? The way I see it, the advantage of going to the Sipura solution is that it is more scalable (ie. I would only need maybe 5 in the beginning and then add one by one as the needs grow) and seems to be cheaper: ~$800 for 8 Sipura's versus $1300 for 1 Audiocodes. The

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

My current config: pstn -> audiocodes fxo gateway -> asterisk -> xlite every fxo ports are registered with asterisk I have this extensions.conf exten => 111,1,answer exten => 111,n,dial(sip/fxo1) exten => 111,n,hangup If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a phone no and connect to the called party. this is a two stage

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

Hi All, Channels 25-28 on a customers PBX are regular Zaptel FXO cards that are hooked into 4 incomming phone lines. They are all in a group to do automatic rollover for outgoing calls (if channel 25 is being used, dial on channel 26, etc.). Sometimes when a user is dialing a number, instead of bridging to one of the FXO cards it goes and rings to Zap/1-1. This doesnt occur all the time but some

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.