similar to: Asterisk and Alcatel digital phone's

Hello, I consider buying three GrandStream GXW4024 and connect 72 analogue phones to asterisk Do you have any feedback how well it works with Asterisk ? I am on a budget, do you have other recommendation for similar setup that get into same budget - connect around 70 analogue phones to asterisk. Thanks in advance. Peter

Hi, In this http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375dating from 2008, experiences with Grandstream GXW4024 were asked. Has anyone something up-to-date to share about this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091005/ad69fa22/attachment.htm

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

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

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:

Hello, For a prospective customer, I need to evaluate the cost of an Alcatel OXE IP trunking licence price. The setup would be: PSTN -----<E1>---- Alcatel ----<SIP>----Asterisk ---------- SIP Phones At the moment, only a couple of simultaneous calls are needed between Alcatel PBX and Asterisk but later, this number will grow.

Hi guys, I need to interconnect an asterisk server (1.4.17) with an alcatel pbx, because i want that an asterisk extension can call an alcatel extension and viceversa. I need to do this throug a NT port (of the digium card) for technical reasons (linked with alcatel hardware). Card installation was fine, but i'm not sure of the configuration i choose. In /etc/misdn-init.conf i put my

Hello everybody I have an asterisk with an integration of alcatel pbx, by sip trunk, all calls are fine, but tha calls calls that originate from a analog line, the recipient is not listening, and that if they hear the call originates, the lines are E1 in alcatel pbx. When a asteris user call to analog line the call is ok. Everyone, has been that problem? I change asterisk version 1.4.21 to

Hi I just came accross this Newbridge Mainstreet 3624 but the Alctel site appears to have zip for reference/user manuals Anyone by chance have 1 of these or a url for the docs ? thx

Does anybody has experience with the SIP phone of Alcatel the ST280. I can't make a call with this phone. Everytime I make a call I get the error Jun 12 19:38:38 WARNING[1133718080]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call 5624@81.124.191.47 for seqno 1 (Response)

Hello list. I'm going te explain my trouble. I have my asterisk with a TDM400P with 4 FXS channels. Two ports are connected to a Panasonic PBX (it's working fine), and others two ports are connected to an Alcatel 4200 PBX (but it doesn't anwer). I connected to a CO port (where i had a pstn line). When I call to the Alcatel PBX, the asterisk show me in it console that es ringing but

Hi there! I wan't to connect my * to an Alcatel 4400. Does anybody have some experiences with that? I have the problem that I can dial in to *, but not *->A4400 . :-( Thx Marc

Hello, I have the following problem with HFC card and Alcatel PBX 4400: - I want to connect Alcatel PBX and Asterisk with HFC card (zaphfc driver 1port S0/T0 interface card) - BRI line is configured at PBX to T0 trunk - In asterisk I have loaded zahfc driver in NT mode and also set in zapata.conf NT mode, point-to-multipoint. - I am using cross "isdn" cable to PBX -

Hi, Does anyone have any experience connecting asterisk to alcatel 4200 series pbx with bri cards? Does it should work with asterisk bri in NT mode, and alcatel bri with TE mode? Cheers, Igor Neves.

Greetings list, I've been asked by someone to help them set up a SIP link between an asterisk system and an Alcatel OmniPCX (v6 software). The asterisk bit's fine, but I know nothing about the Alcatel except that it does apparently allow the setup of SIP trunks. Does anyone have experience with the Alcatel unit? Any obvious pitfalls to watch out for? Any suggestions gratefully

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new flags that need setting, etc? Thanks Jason -------------- next part -------------- An HTML attachment

Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp (bhcompile@decompose.build.redhat.com) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps on digium website...no errors reported. The modules seem to have loaded...here's what lsmod shows: Module Size Used by