similar to: IAX IP Phone

Hi All; Anyone can advise for a good IP Phone that has the ability to support SIP firmware and IAX firmware? Ofcourse, SIP there is a lot, but we need also the ability to use IAX (as it is good for NAT). Any advise. Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo!

Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal

Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg -- Kyle Sexton

Hi All; Can I do transfer for the call from zap/1 to zap/2 (both are fxs)? All what I need is to add the t argument for the Dail function? And how can I transfer to be in that senario: zap/1 dial a code to transfer for zap/2, once zap/2 answered, then he can talk with zap/1 (where the third party will not hear what they are talking), then if zap/1 hanged up the handset, the call will be between

*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards

Hi, Wrong UIDs are returned from mailbox_transaction_commit_get_uids() in dovecot-1.1.rc3. The problem is in: int mailbox_transaction_commit(struct mailbox_transaction_context **t) { uint32_t tmp; return mailbox_transaction_commit_get_uids(t, &tmp, &tmp, &tmp); } It should be: int mailbox_transaction_commit(struct mailbox_transaction_context **t) { uint32_t tmp1,

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

Hi. I'm a dovecot newbie. I've set my pop client to leave messages on the server for 7 days. I'm downloading the same messages over and over as dovecot appears to be overwriting X-UID's in the mbox's. I can't think of any other program on the system that might write to the mbox other than postfix (and that program doesn't appear to be writing X-UID). My setup has one

Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in

Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----

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 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 would like to know if the following misdn warnings are relevant. Currently, I don't need echotraining. However, I took a quick look at the * source code and l1watcher_timeout seems to be defined (echotraining was not found). Currently I'm setting l1watcher_timeout to 0 which is default (so I suppose that this warning won't affect me). Any comments on this? *CLI> misdn reload

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

Hi I have two users on one asterisk server A(192.168.1.100) while the other two users are registered with asterisk server B(192.168.2.50) and i have configured a PC to act as a router between two subnets(Ping and other applications are working). How i should configure those two asterisk servers so that the users from different subnets can communicate with each other thanks in advance for your

Greetings, I'd like to set up samba to use group and user credentials from our windows domain in mixed mode. I want to be able to give windows users rights to samba file shares without having to set them up with a Unix account locally. If a user does have a local Unix account I'd like to be able to use the local account for permissions. Is this possible with winbind? I've set up

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Hi R-users, I would like to find the goodness of fit using Chi-suare test for my data below: xobs=observed data, xtwe=predicted data using tweedie, xgam=predicted data using gamma > xobs <- c(223,46,12,5,7,17) > xtwe <- c(217.33,39,14,18.33,6.67,14.67) > xgam <- c(224.67,37.33,12.33,15.33,5.33,15) > chisq.test(xobs, xtwe = xtwe, rescale.p = TRUE) Error in chisq.test(xobs,

Hallo! <?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p> <o:p> </o:p> I want to compare different samples with the formula … <o:p></o:p> <o:p> </o:p> Here some example data: <o:p></o:p> <o:p> </o:p> <o:p> </o:p> sample1<o:p></o:p>

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.