similar to: Connecting Analog to SIP gateway to asterisk server

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send "qualify" queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Regards --------------

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Selling analog GSM gateways at 119.00 USD. Polarity reverse for "call answer" and "call end". Compatible with all Digium/Asterisk analog FXO cards. Suitable for low-traffic call termination, inbound calls from cellular network or for phone bill optimization applications. Main Functions - Provide Accurate Anti-Polarity Signal - Incoming Calls Display - Add Prefixal

Hi all, Anyone ever used GoIP GSM SIP Gateways ? If yes, what was your experience with those ? I'm looking at this: http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBX&hash=item415d37377c If anyone has any (good) experience with another brand, I'll take the names and models. Thanks

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

Hello, There are some parameters in the ISUP data (coming into the network via SIP-T packets) that need to be translated into SIP headers to be used by asterisk for proper call routing. What gateways are available to accomplish this? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:

Searching through wiki and google. http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html but there are also other products on the market. --- Wondering if its possible to connect as follows: Extension -> Asterisk -> ZyxelAnalogTelephoneAdapter -> GSM gateway. The best way would be to make the ZyxelAnalog.. to be a channel. But I don't think that is doable.. or ? ---- So i

I know it is a strange arrangement but due to contracts, it is what it is, no PRI for now. I wonder if anyone on the list has run a server with both types of cards installed? Results? I have never touched a BRI except in concept and Cisco lab. Not sure what the BRI stuffed package may or may not do to anything else that might relate to zaptel or Sangoma. Thanks, Steve Totaro

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

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

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

Hi, Ive been searching the lists and cant find the exact solution I need using *. I need to route voice channels between two sites across cisco routers.Both PBX's are analog only (no Digital upgrade path). I was thinking for the Gateways at each site.Have 2 FXS/FXO in each gateway.Two ports can be connected to the PABX incoming tie lines and two connected to extensions.LCR can be

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

Hello everyone, I am pulling my hair here because a carrier threw me curve early today. They want to send calls to my asterisk server using SIP. Then they said that their gateways don't have to register with my server, that all they have to do is send a prefix for validation. Whereas I can think of several ways to authenticate their incoming number string, I am only used to the orthodox