similar to: OT: Upgrade Addpac AP200C

Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any

I am trying to connect an ADDPAC 200 to * using SIP protocol. Problem is the documentation for the ADDPAC is not very helpful (they don't mention SIP at all, seems to be a newer firmware). I think I got the * part working but the ADDPAC setup is the problem, I will appreciate if somebody can post a working example or point me to documentation in english (found stuff in russian and korean so

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

Hello, At a client site yesterday I installed a dozen GrandStream GXP-2000's with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX and phones: network access for users windoze PC's through the phone's switch port was unbearably slow, making it almost impossible to work. When plugging back PC's directly to the LAN speed was normal again. On my test setup

Hi all, Today I've stumbled upon a very strange behaviour with an analog fxs/fxo gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html) connected to a CVS HEAD(from today) Asterisk server. This manifested itself after enabling the CallerID on the pstn lines connected to the FXO ports of the module. Both FXO modules have their own sip username/passwords and are registered to the

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

I am trying to transport a dd image between to hosts over a cross linked gigabit connection. Both hosts have an eth1 configured to a non routable ip addr on a shared network. No other devices exist on this link. When transferring via sftp I received a stall warning. Checking the logs I see this: dmesg | grep eth e1000e 0000:00:19.0: eth0: (PCI Express:2.5GT/s:Width x1) 00:1c:c0:f2:1f:bb

Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with

Hi. I'm new in Asterisk with MGCP. I set up a MGCP user agent and start asterisk with the next configuration files. '--------------- extensions.conf ---------------------------------------------------- [general] static=yes writeprotect=yes [globals] ap1 => mgcp/aaln/ap200@64.76.148.186 [macro-apl1] exten => s,1,Dial(${ARG1},30,Ttmr) ;exten => s,2,Voicemail(u${MACRO_EXTEN})

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

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Good Morning, I would like to know if is possible to do a conference with 9 client with asterisk. The client is connecting to sever through lan, we think don't use PSTN or ISDN. Thanks, Alberto -- Alberto Carlana <alberto.carlana@virgilio.it> -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes

Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hello, We have been using a Diva 4BRI with our Asterisk PBX through the capi interface for almost a year now with good results. However, recently we started to hear heavy clicking sounds in our phones when two simultaneous incoming calls are processed by the card. The clicking does not originate with the phones as it happens also in voicemail left directly on the server and happens with different

https://bugs.freedesktop.org/show_bug.cgi?id=101220 --- Comment #12 from Ilia Mirkin <imirkin at alum.mit.edu> --- (In reply to Pacho Ramos from comment #11) > I don't use any initrd... I could rely on CONFIG_EXTRA_FIRMWARE... but I > thought I didn't need that as nouveau is compiled as a module and not into > the kernel :| OK, well if modules are loaded off the FS and not

https://bugs.freedesktop.org/show_bug.cgi?id=101220 --- Comment #13 from Pacho Ramos <pachoramos1 at gmail.com> --- (In reply to Ilia Mirkin from comment #12) > (In reply to Pacho Ramos from comment #11) > > I don't use any initrd... I could rely on CONFIG_EXTRA_FIRMWARE... but I > > thought I didn't need that as nouveau is compiled as a module and not into > >

Dears I installed digium fax and followed the instruction at http://downloads.digium.com/pub/telephony/fax/README,And as you can see above that t38 is loaded I am using a call file to send fax1.tif file as fax to the gateway named add The problem that Addpac send always Receive 488 Not acceptable here,and lkindly find my debug attached Please advice. Thanks I Advance shark*CLI>

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the