similar to: ATA phones ring when they register

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

Hi, I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The setup consists of the following components: - A Digium TE121 for connectiong to E1 ISDN - Debian box with Asterisk 1.4 - Grandstream GXW-4008 SIP ATA to which the Fax machines connect I am aware of the problems with this type of ISDN <-> Asterisk <-> SIP ATA <-> Fax machine installations, e.G.

Hello list ! I don't have the money to test out all the products and reading the manuals is not always that enlightening... Does someone here know a good gateway-product that lets analogue telephones communicate with an Asterisk-server. I have found the Grandstream GXW-400x to be able to add SIP-accounts to analogue telephone devices that are connected to the FXS-ports. Moreover this

I'm trying to use a GXW-4008 for the first time to provide simple POTS. Is anyone using it? How about samples of SIP.CONF and EXTENSIONS.CONF? Do you have advice for configuring the GXW-400x for this application? How long a local loop will it support on the FXS ports? When I started to configure the unit, I was able to connect via the WAN port. Now I'm unable to connect to

I'm currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and

Dear all, Just wanted to know if any one had deployed the Grandstream GXW 4024 yet. Wanted to hear any feedback and/or problems with this unit that you may have experienced. Thank you. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

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

Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm

HI There is a question about the fxo of the zaptel card which is set a number to use as common analog phone. When I use ${CALLERID(num)}to get it's number, it could'n be done. But ${CALLERID(num)} could get the other number of the SIP or IAX . Could you tell me the reason, and how I could get the number of the fxo which is used as a common analog phone? Thanks

We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. -Thermal -------------- next part -------------- An

Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? kel

Ok, dumb question. On a certain LAMP server I am seeing in 'ps auxf' a process called "unit" with no arguments or other path info. It has a fairly low pid, 3041, indicating it might have been started soon after reboot (last week). but ps says it was started yesterday, I don't see it on any of 3 other CentOS machines. It is hard to google for such a generic name. So does

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

I have following settings done on my Fedora8: Downloaded openh323-v1_19_0_1-src-tar.gz pwlib-v1_11_1-src.tar.gz Extracted them in /root/openh323 and /root/pwlib Exported the following variables: PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Then I compiled pwlib and it was fine. But in

SECURITY QUESTION & SANITY CHECK: If only my SIP ports and a small range of RTP ports are facing the public internet, what is the method by which an evildoer would be able to do fraudulent long distance on my nickel? Would it REALLY be as simple as guessing the credentials for ANY of my local sip endpoints? Like most people, my local endpoint credentials would be easy to guess: Username

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 have a FreePBX system with PRI trunks that's doing a number of things very nicely, but frustrating me in one area. I am using a Grandstream GXW-4008 in an off-premises location to provide "POTS" service on four ports (this device worked fine in an early application using a hardware VPN to the Asterisk server). The Grandstream has a public static IP port, as does the

Dear List, I just installed R 2.6.1 (on Win2K) and I get a strange error in functions min() and max(): > min(1:3) Errore in .Internal(min(..., na.rm = na.rm)) : nessuna funzione interna "min" which, as you may have guessed, means 'no internal function "min" '. The same happens for max(). Maybe this is a bug in the new release, or maybe I'm missing