similar to: Need Adapter/Gateway with PSTN-interface

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

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

After upgrading from 2.0.10a to 2.2.4, I'm seeing many new messages being logged. What does this mean? Jun 5 09:42:50 files2 smbd[27585]: [2002/06/05 09:42:50, 0] smbd/oplock.c:oplock_break(788) Jun 5 09:42:50 files2 smbd[27585]: oplock_break: no break received from client within 30 seconds. Jun 5 09:42:50 files2 smbd[27585]: oplock_break failed for file

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Does anyone have any recommendations on an IAX Desktop Telephone or ATA Device. I currently have 2 of the SIPURA-841's on my local network and now I am wanting to try an IAX Device at my remote office since I think that it would be easier to configure through various routers than a SIP Device. I just started to look at the Digium IAXy Single FXS

I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is no on-hook state in the whole setup. Obviously If I just connect the input to a port on an ATA, I'll just get a dialtone played through the speakers. Can anyone think of a way I can

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

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 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.

I've heard now from more than one source about problems with CentOS (and RH) at least up through 5.2 w.r.t. SATA drive handling, and I've even reported on this myself in this list before. My question is, do we have any idea if 5.3 has any improvements in this area? One of my cohorts here, who happens to be a Fedora fan, says that these problems are fixed in F9, but I have grave concerns

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

Hi, We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2 extensions to iaxmodem devices for fax2email. We are rapidly growing and will be adding an additional PRI trunk and grow to about 150 SIP & IAX2 extensions towards the end of the year. We have two Digium Wildcard TDM800P cards (8 x

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.

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

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 I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

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

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can

While debugging stale backgroundrb it became apparent that there''re memory leaks in it. Here''s top object counts dump: 36715 String 6211 Array 5598 Proc 3150 Hash 1478 DRb::DRbObject 1313 Regexp 1272 Class 750 Thread 741 Socket 739 Slave 739 Slave::LifeLine 737 Process::Status 691 DrillReminderWorker 391 Module 160 Range 75 Gem::Version

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