I'm having speaker phone echo issues with my grandstream phones 100.i understand that the echo'ing issue is only obvious because of the round trip latency and that traditional phone lines have echo's too but because there is such a slight delay, it can be mistaken for side tone, which is
PROTECTED] wrote: Christopher Corn wrote: May I ask, from your own personal experience. is it not necessaritly worth (the headaches) of investing mytime into setting up SPANDSP into my asterisk system, but rather invest it into going to a company, like packet8 that offers t38 conversion?I am
one that also offers support for it. thanks.___
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I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for now i will have to setup my fax machine to connect directly to my t38 provider. anyone know when trixbox may have this update?___
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I've been reading about FOIP and there something i dont undrestand. maybe someone can explain to me.A couple of faxing methods im confused about.The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversionCan someone explain to me why the pass
to be where i should go.Thanks. Lee Howard [EMAIL PROTECTED] wrote: On Sun, Sep 24, 2006 at 01:58:21PM
-0700, Christopher Corn wrote: A couple of faxing methods im confused about. The pass through method, sending fax data over G711 codec versus Relay method, t30 to t38 conversion Can someone
I help support a small office, 5 SIP phones, connected to an Asterisk PBX. We have 4 analouge fax machines connected to a pstn that i would like to get rid of, but need a foip solution. rather thing trying to do a pass through using the g711 protocol, I want to go with a t38 termination since
My Asterisk PBX now is strictly IP, out a 6000/600kbps using the g729 codec. Only thing using pstn now are my fax machines.Thanks for the input Jay. "Jay R. Ashworth" [EMAIL PROTECTED] wrote: On Sun, Sep 24, 2006 at 08:06:41PM -0700, Christopher Corn wrote: I help support a small
I have an trixbox system setup, and all my phones are IP based, different Grandstream phones to be exact. I have some fax machines that are still around, using analog lines of course. I've read a little into FOIP and the changing of the signals form t30(traditional fax machines) to t.38 ( ip
Im using voipestreet and voxee for my SIP termination. neither of them, offer any kind of e911 service. as i search the web i see different companies that offer this e911 service to voip suppliers. I want to choose the right one, seeing how in an emergency, it can be very crucial. any suggestions?
This is what I did, i installed an the open source codec. but legally, your supposed to buy a license.1) download 723 and 729 codechttp://kvin.lv/pub/Linux/Asterisk/2) copy both codecs to /usr/lib/asterisk/modules3) restart asterisk/etc/init.d/asterisk restart4) verify its working. as long as
hmm, no we have about 7000 employees.Michiel van Baak [EMAIL PROTECTED] wrote: On 23:47, Tue 19 Sep 06, Michael Neuhauser wrote: On Tue, 2006-09-19 at 13:45 -0700, Christopher Corn wrote: michael, at my real job, the phones display peoples names when calling out from your phone. how
i have trixbox running, the latest version and when i make an outgoing call from this phone it doesn't pick up the user's name, but instead display the number. is this a grandstream problem? or asterisk? i did specify the user name from 'extension" within trixbox. in matter of face, if i call
not sure. thanks.Marco Mouta [EMAIL PROTECTED] wrote: test it with someextension and using a sipphone like xlite or something else and you will be able to understand if it is a grandstream issue!hope it helps On 9/19/06, Christopher Corn [EMAIL PROTECTED] wrote: i have trixbox running, the latest
michael, at my real job, the phones display peoples names when calling out from your phone. how is this done?Michael Neuhauser [EMAIL PROTECTED] wrote: On Tue, 2006-09-19 at 11:53 -0700, Christopher Corn wrote: ... i dial now from SIP phone to SIP phone. one being a grandstream gxp grandstream
im sorry if im not being clear. when im calling from my gxp to the grandstream 100.the gxp doesn't pullup the users name from grandstream 100.someone in another mail mentioned that this is not the way its supposted to work.my office does this, when i dial someones number, it displays their
Anyone have a procedure on how to get caller id to work with trixbox? theres a procedure for it to work with asterisk, but not asterisk. thanks alot.___
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call waiting was then enabled. it seems that one step in configuring is to enable call waiting, as it appears to be disabled by default.Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Sep 12, 2006 at 09:55:17PM -0700, Christopher Corn wrote: Christopher Corn <[EMAIL PROTECTED]>wrote: i'
i finished setting my asterisk pbx with 5 phones. thanks for everyones help here, in getting this accomplished. it is greatly appreciated. this is what i set up.Athlon 3500+ cpu (2ghz i think) 1 Gig of RAM Netopia gateway to SBC DSL 6000kbps/600kbps Linksys wire with QOS functionality
im finding companies like voxee that offer very low rates and then companies like voipstreet that offer at a higher rate doube. whats the catch? is voxee, what you would call a wholesaler and voipstreet, commercial?im worried about going with companies like voxeee, because i question their
i've got trixbox installed and grandstream 101 phones. out of my 4 phones, one of them has call waiting working. they all the same version of firmware and settings. i tried looking in asterisk to see if anything could be doing this, but can't find anything. suggestions?
nevermind i figured it out :)Christopher Corn [EMAIL PROTECTED] wrote:i've got trixbox installed and grandstream 101 phones. out of my 4 phones, one of them has call waiting working. they all the same version of firmware and settings. i tried looking in asterisk to see if anything could
i found this out also, after purchsing the phones.Tom Vile [EMAIL PROTECTED] wrote: They only do numeric callerid. On 9/11/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: I have tested "Grandstream Budgetone 102" and "Grandstream Budgetone200" and with both, if they are called from a caller that
positive? negative? yay? nay?___
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i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats possible, without tying up a
rich, thanks for replying. i assume your talking about enabling call forward and call forward on busy from my vsp side. i dont quite grasp everything else that your saying, can you explain in laymen terms. thanks.Rich Adamson [EMAIL PROTECTED] wrote: Christopher Corn wrote: i plan on buying 4
okay, i undrestand what you guys are saying. thanks alot.Brent Franks [EMAIL PROTECTED] wrote: i would think i would need one DID with multiple simultaneous connections.Hello, you can't setup a DID per se on an analog line.Essentially what you want is 4 regular POTS line in a hunt group.The
all thanks for the replies. i know what to do now. thanks.John Novack [EMAIL PROTECTED] wrote: What provider?Pots lines?SOME providers will provide hunting on residential lines, but not all, and most probably not 4 lines. Hunting does not require any thing more than the providers switch
im having issues when routing calls from the outside with my new VSP. this is what asterisk tells me when i try to make an incoming call, i get the no service response when i call. -- Executing GotoIf("SIP/christopher_corn-eddb", "1?from-trunk||1") in new stack -- Goto (from-trunk,s,1) --
can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones)and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for
understand the terms, and just offers them what used to be called a "phone line" - the ability to make calls (termination) and recieve them (origination). i hope this helps,
yair On 9/10/06, Christopher Corn [EMAIL PROTECTED] wrote: can someone please explain the differnces
offers unlimited calls, in and out in the US asterisk support no setup fee and support 729 codec? and of course is reliable and clearthanks alot.___
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thanks for the verbose explanation!Rich Adamson [EMAIL PROTECTED] wrote: Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers
ok maybe thats asking for too much. how about a voip provider that provides 729 codec support ? :)Christopher Corn [EMAIL PROTECTED] wrote:offers unlimited calls, in and out in the US asterisk support no setup fee and support 729 codec? and of course is reliable and clearthanks alot
I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled.is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because
i see. thanks for the info.[EMAIL PROTECTED] wrote:Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.-- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned
] wrote: Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.-- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use
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