similar to: compare Linksys SPA8000 and Grandstream GXW4008

Hi, I have a set of Grandstream GXW4008 (units of 8 FXS ATAs) and another set of Linksys SPA8000 (8 FXS ATAs). The GXW4008 has a "neat feature" called "early dial" which allows me to define a "dial pattern" as generic as {*X+,#+,X+} (or something similar; the idea is to "match all digits") and send those digits >>immediately<< as they are

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's identical to the PAP2 but with 8 ports instead of 2) and the telco negotiate with each

Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm

Hi guys, Anyone have experience using the Linksys spc tool to provision devices? I've been fighting with it all afternoon and can not get the tool to parse a xml file properly. I can get it to generate me a .cfg file from a text file, but when I tell the unit to pull down the cfg file it doesn't really look like any of my settings were even set from the file I generated. Thanks for any

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]:

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

I have an Asterisk running with both GXW4008 (FXS) and GXW4108 (FXO). The FXS Gateway works perfectly, no problem so far. The FXO Gateway (GXW4108) also works fine. The configuration for local settings in Brazil was quite easy, however, I still not able to make Caller ID to work. I'm setting as DTMF Caller ID type, but still not working. Let us know what kind of problem you have, maybe I

I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive

Hello, I want to ask that if thee are some ATA decives that i can use to connect mutliple analog phone lines to my VOIP system.. I mean for example an ATA device with 24 ports with 24 independent SIP accounts. For example for some dormitories in my area, i want to put an ATA device and move existing lines to VOIP accounts. Only problem is, if i dont give seperate SIP accounts for all ports, i can

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;) I wonder where I can buy 50 pin Amphenol cables, with connector on one side, and open cables on the other for mounting in our own patch panels. In Europe, or Sweden preferably. It's said to be very common on telcos, but I can't find it for sale anywhere! Thanks!

Hi, I just got a T1 interface for a AT&T (became Lucent) System75 (uses same cards as Definity). I would take a crack at making a cable but can't determine the pinout for a cable and it is not apparent from the board. Asking you guys makes sense as one of you may have one of these systems. The cable has a amphenol male D50 connector on one end and probably a rj45 on the other. I also

Hi all, I've got a few devices, SPA112's and SPA8000's, that are giving me problems. Each device has a separate SIP credential for each port, but sometimes, only a few of the ports register. So, the device will be running fine for a while, then suddenly one or more of the ports will become Unreachable. These ports will stay unreachable until the device is power cycled.

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks.

We have been experiencing static noise on one of our phone lines connected to our Adtran Total Access 750 which is then connected to a T100P card. I was convinced that, after rain storms, the static would get worse, it was surely a problem of Verizon's. Verizon sent a field rep out today and he told our secretary (I was busy) that there was a hard ground on the Ring portion of the line.

Does anyone have any experience using asterisk with an external predicitve dialer, like MediaTel? Specifically: The predictive dialer dials out over T1 circuits. It connects to asterisk via amphenol cable from an fxs card in the dialer to asterisk with a tdm2406 fxo card. In the analog world, the dialer dials out through the t1 circuit, and the fxs card is plugged into a 66 block so the

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the