search for: addpac

Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can load addpac. Is there anyway that can I upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the...

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...

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...

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...

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. Office...

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

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 RFC3665, clause 2.2 (http://www.zvon.org/tmR...

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

...0 200 OK Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport=5060 From: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7 To: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4 Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Contact: sip:5555832351@xxx.xxx.148.242 Content-Type: application/sdp Content-Length: 207 v=0 o=5555832351 1114766125 1114766125 IN IP4 xxx.xxx.148.242 s=AddPac Gateway SDP c=IN IP4 xxx.xxx.148.242 t=1114766125 0 m=audio 23268 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000/1 a=fmtp:10...

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

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> fax show capabilities shark*CLI> Registered Fax Technology Modules: Type : T.38 Description : Digium Fax T.38 Driver Capabilities :...

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