similar to: Jumpers and DIP switches on Atcom AX-4S and AX-1E

Hello I'm thinking of selling an Asterisk server based on Atcom's IP02 solid-state unit with one FXO and one FXS ports: http://atcom.cn/En_products_IP02.htm By default, this unit based on a 400MHz Blackfin 532 chip only has 64MB RAM and 256MB of NAND flash. Those can be increased to 128MB and 1GB, respectively. Do you think I can install Linux + Asterisk + LAMP (replacing MySQL with

Hello According to this article, this nice little unit can only use the PSTN port for outgoing calls (ie. as a backup in case the connection to the VoIP provider stops working), but not incoming calls: http://tinyurl.com/mwjmo8 Can someone confirm that Atcom made this strange decision, and that there's no work-around? Thank you.

Hello All We purchased *25* new AtCom AT- 530 phones. Four of them did not work for even once and some of them lost configuration after some days. I talked to AT Com people over chat for support. They have just 1-2 people for support who are also busy in some other activities due to which they are unable to communicate in proper way. Often support guy left conversation suddenly and is unavailable

Hi, I have a few of Atcom AT320 phones. I can choose if install SIP or IAX2 firmware. Any suggestion? Mimmus

Anyone here used these phones? I'm getting more and more frustrated by todays modern crop of routers with their so-called SIP ALGs which are invariably broken, or routers with built-in ATAs which block internal SIP phones from working, so looking to use IAX for some end-users. I already support it for people who want to use (eg) Zoiper and use IAX a lot to plumb boxes together, but never

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

Hello everyone, I have a Atcom AX-1600P(1) card with a FXO module and I can't configure it. I have four extension with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

Hi all, Diax version 0.9.9f is ready to be tested by the interested people. You can download it for the moment from the following location only: http://www.geocities.com/tdanro/diax/diax099f.zip Please do not use older config files with 0.9.9f !!! You have some command line options now for diax.exe: /d - start with debug mode enabled /u - start with ATCOM USB phone support enabled

I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps someone on the list has experience with this. Is there a way to get MWI support for PA168V-based ATAs? Apparently some IP phones based on the PA168V chip has this support already (Atcom AT-320 for example) by configuring Asterisk with 'mailboxdetails=yes' in iax.conf. On my ATA, however, it does nothing. Any

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

Hi all! I'm new with telephony cards and DAHDI drivers. I have installed Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. I'm following the installation guide of Atcom [1] for AX1600P analogic card, modules are loaded [root at pbx ~]# lsmod | grep -E "hisax|netjet|dahdi" netjet 14618 0 isdnhdlc 4523 1 netjet mISDNipac

I'm looking for an ATA\Voice Gateway that runs IAX and has several ports (8 would be nice). I am looking to avoid devices that use the same firmware as the ATCOM devices as I found them to be buggy (and a PITA to find the proper update). ---------- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed...

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the

Hi, I just got a value for the dip test out of my data of 0.074 for a sample size of 33. I'm trying to work out what this actually means though? Could someone help me relate this to a p-value? Thanks James

This could be a nice opportunity for users with a high volume of SIP traffic terminating in the US: Collecting dip fees on outbound phone calls - fees that would otherwise go to the local phone company. With all the recent fees and surcharges, the cost of wholesale telecom and dialer traffic keeps rising. But what many companies with a high volume of IP based voice traffic don't realize

Hi all, I am using Hartigan & Hartigan's [1] "dip test" of unimodality via the diptest package in R. The function dip() returns the value of the test statistic but I am having problems calculating the p-value associated with that value. I'm hoping someone here is familiar with this process and can explain it. In the original article there is an example using n=63 and a

For those of you who may be interest.... IAX2 loads are now available for the standard builds... http://www.aredfox.com/edownloadsiax2.htm Just a word of caution... Remember to change the ports over to 4569 from whatever. And don't forget to grab the palmtool from http://www.aredfox.com/download/tools/PalmTool.zip My own testing of IAX2 with both a phone and an ATA is that IAX2 is