[asterisk-users] transfering calls
Hello folks! I've added a tT statement in my extensions (softphones and hardphones) so both the caller and calling parts can transfer calls (the strange thing is that some extensions end up only transfering the calls they originated, even if the config is the same of all other working extensions, which are able to transfer both placed and received calls). However, my trouble comes with the PSTN. I've added a t on the POTS- incoming s extension, so when it dial an extension, that extension can transfer the PSTN to someone else. Similarly, I've add a T to the outgoing calls Dial, so I can transfer a call to another extension after I place it. However, what happens is this: when I transfer a PSTN call to an extension, the PSTN gains the ability to transfer it! How I can prevent that to happen? Cheers, Francis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with Sipura SPA 3000
On 11 de abr de 2007, at 21:07, James Harper wrote:
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply the dialtone does add (a
slight) additional load, rather than it just routing calls between
endpoints. Not an issue with one or two ATA's though.
i have just one ATA anyway, this is intended to be used solely at
home... I'll probably give it up in favor of pbxes.org...
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for
you.
From the 'sip_extension_in' context you can make a single '0' or
'*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
A few things to check:
. ${EXTEN:1} will be empty because the extension can only be '0'.
Change
it to 'SIP/LinkSysOut' instead
Done.
. I'm not sure but I think that the SPA3000 can either present a
'false'
dialtone to the SIP call on the PSTN line, take the digits, then send
them to the PSTN then connect the SIP call to it, or it can give the
real PSTN dialtone and connect the call immediately. I think the
latter
is what you want but I can't remember the name of the setting. Maybe
'one stage dialling'?
Done It works! I had to disable one stage dialing and setting the
VOIP DP to none. However, this is giving me one trouble: I use also
my cellphone (E61) to make calls, and it would be nice to do one
stage dialing with it. I don't think it's possible to make it one
stage with the mobile and two stage with the FXS of the Sipura...
Cheers, and thanks a lot!!!
Francis
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[asterisk-users] Asterisk, nat, gizmo and fwd
Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda worked, but it wouldn't receive any calls. The Gizmo one worked ONLY to call their echo test - If I tried to call a real number I got some serious noise, sometimes the phone ringing, but never getting audio through. It also brings my asterisk down (1.4.0) - it just hangs, without the peers managing to authenticate on the server again. Has anyone succesfully managed to connect to Gizmo and FWD in a similar setup and would be so kind to share the configs? []'s Francis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote: I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. Get the impedance settings right. An impedance mismatch will cause echo (but may not be the only cause) Thanks a lot for your answer!! But, how do I found out what's the correct impedance of lines here? But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. I think the 'echo suppression' setting causes this. It is meant to reduce the incoming audio (and hence the echo) while you are talking, which can be annoying but is supposed to be less annoying than the echo itself. I see... 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! A dialplan of '(S0:s)' will get your phone to jump straight into the 's' extension in asterisk as soon as someone picks it up. From there you can do something like: [sip_ata_incoming] exten = s,1,Answer exten = s,n,DISA(no-password|sip_extension_in) so Asterisk will give you dialtone and do the dialplan stuff for you. From the 'sip_extension_in' context you can make a single '0' or '*' call the PSTN line. I think if I choose the * to get a dialtone it won't work because it seems that the SPA-3000 will pick up that character and use it as if I was trying to access its own services... By the way, for transfering calls, will asterisk or the SPA the one that will actually do the transfer? Good luck with the echo situation. I have an spa3000 and no matter what I do I get echo coming back to me with almost no reduction in volume!!! Thanks... I don't mind if the echo is small, I actually prefer a small echo than that cutting thing... :( Cheers, Francis James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
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[asterisk-users] help with Sipura SPA 3000
Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect to Asterisk
On 4/10/07, Mike [EMAIL PROTECTED] wrote: Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone My own Asterisk - Outside lines But when it comes to smaller villages (I deal with people in tiny places), I'd like to reuse their own PSTN line this way: Customer's SIP hardphone My own Asterisk -- Some device on the customer's premise customer's PSTN lines I know ATAs are mostly used in a scenario where you reuse traditional phones to connect to SIP servers, but can they accomodate my scenario? And if so, what line of ATA should I be looking at? Mike Hello Mike, Wouldn't a Sipura SPA 3000, with an FXS and an FXO, handle what you want? Cheers, Francis -- Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk without PSTN interface cards
On 4/10/07, Alejandro Cabrera Obed [EMAIL PROTECTED] wrote: People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with apt-get the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards, Hola Alejandro, I used asterisk some days on a mac without any PSTN whatsoever, just to talk between softphones (and ip phones). No problem with that. Cheers, Francis Francis Augusto Medeiros ICQ:7825595 Skype: francisaugusto AIM/iChat: francisaugusto Vitória da Conquista - Bahia - Brasil Lâmpada para os meus pés é a Tua palavra, e luz para o meu caminho --- Salmo 119:105 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External BRI - is there such thing?
Hello folks! I wonder if, the same way we've got external FXO's (SIPURA's and the like), there is the possibility of having external BRI's so ISDN lines could be connected to Asterisk via LAN. I am considering running Asterisk on a MacOS X machine, and, even if I opt for Linux on an i486 machine, I'd still like to go external. Any comments on this would be higly appreciated. Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] KS-Like controlling
Dear friends, Is there any way to let users know if lines connected to Asterisk (via FXO or BRI) are free to be used? I mean, a KS-style thing. I wonder if it's possible to do something like that via web, or, even better, to display such info on ip phones. Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Thanks a lot Daryl!! Yours, Francis On Wed, 18 Aug 2004 10:25:21 +0930, Darryl Ross [EMAIL PROTECTED] wrote: Hi Francis, so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) Actually, we also have non-fixed phone numbers in Germany. I think this is not weird, I think this is very good. And again, Asterisk supports this. Oh, so I how does Asterisk knows when to start dialing out the numbers, if there are no rules? Have a look at http://www.voip-info.org/wiki-Asterisk+Extension+Matching Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hallo Holger, and thanks for your explanations! Here's my reply: On Mon, 16 Aug 2004 11:51:54 +0200, Holger Schurig [EMAIL PROTECTED] wrote: Well, I'm not really looking for a lot of phone features, just the basics (transfers, call retrieval, etc.). Not sure what you mean with call retrieval. But the GS phone can only do blind transfer. By call retrieval, I mean this: when the phone rings on an extension (incoming call), but I'm far from it, then, dialing a certain prefix would make me pick up that call from the extension that's nearby me. Oh, bummer, if the GS phone can't do attended transfer, than it's no good for me. I thought that all this could be done just by asterisk, not needing a supporting phone for that. Also, Asterisk currently can only do blind-transfer by software. Really??? An attended transfer is a MUST for me. What are my options?? If you want attended transfer or supervised transfer (two names for the same thing), then you can't currently use GS. I'll most likely use a BRI. Do you think this will help to avoid echo? I heard that you can get echo with chan_capi as well. If you get a HCF chip based BRI card and use zaphfc with chan_zap, then the echo cancel code in zaphfc can kick in and remove echo. That said: I had never echo with neither chan_capi nor chan_zap/zaphfc. Well, being from Germany, you probably know that Teles.isdn card for PC's. Do you think that thing is any good? I've only seen it in Norway in 1999/2000, and that's the one I got spare here. Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Mon, 16 Aug 2004 11:44:53 +0200, Holger Schurig [EMAIL PROTECTED] wrote: My concern was if I'd have to teach folks how to dial, but I guess that I can still have the option to assign a number that will give immediate access to the PSTN, In Germany, you usually use a 0 in hardware PXSes to get the PSTN dial tone. No problem with Asterisk to do the same. Same here, that's what I want to do. so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) Actually, we also have non-fixed phone numbers in Germany. I think this is not weird, I think this is very good. And again, Asterisk supports this. Oh, so I how does Asterisk knows when to start dialing out the numbers, if there are no rules? This is really a great idea. See, my biggest concern is not the voice quality in terms of audio, but if the conversation is allowed to flow in the same way as with regular phones. For me this is not a problem. I tested two different PA168 based phones, a Sipura SPA-2000 and two Grandstreams BT101s. Just the PA168 had noticable delay. Botht the Sipura and the Grandstreams had analog-phone-quality. I'm talking here from phoning inside my network, so there were no internet delay etc. The PA168 based ones where slower. Hmmm, seems interesting. I'm growing to opt for this Sipura thing. To avoid echoes, I'll opt for a BRI instead of any FXO available out there. The used code has an implication to the delay. Because I use the phones inside my network only, I opted for alaw or ulaw, because they are faster. I don't care for good compression. Make all phones using the same codec, then Asterisk won't need to do any codec-conversion, this saves a millisecond (or so, see show translations) as well. That wil be my case as well. I also used both chan_capi and chan_zap with zaphfc to phone to and from EuroISDN lines as FXO. Again there was almost no delay and no echo. Never tried analog FXO. Cool! So I think this setup would be ok for interrupting Brasilians. :-) eheheh we never know, we can be VERY interrupting... ;) PS: search at www.voip-info.org if you don't know what I mean with PA168 I will! Thanks! Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 12:15:20 +0200 (CES), Tobias Jönsson [EMAIL PROTECTED] wrote: On Sun, 15 Aug 2004, Peter Svensson wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Not necessarily! When you call an analog phone via isdn, the other end will introduce echo so that the ip side will be hearing himself speaking with a small delay. I have that problem with my home BRI running zaphfc. Regards, Tobias Jönsson, Lund SE Hej Tobias, Is this small delay annoying enough? Can it be perceived by the part at the pstn side? Does it disturb fax signals, for example? Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:03:49 -0300, Nicolas Gudino [EMAIL PROTECTED] wrote: Hi Francis, If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. The thing is that it's a new office, so we can choose what kind of wiring to use... My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. Internal calls (SIP to SIP) will sound great. You will probably experience some echo when going to POTS. I did not try the Sipura SPA-3000 yet, but it seems to be a cheap alternative to a gateway, providing you with one FXO and one FXS for $130 or so. the echo cancellation in the sipura works well for fxs, it might work well to for fxo. Gracias Nicolas! I'll really give it a look... Too bad that with this option I'll loose the LCD's, but, what the heck... ;) Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:39:10 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: Why on earth would you install SPA-2000s and endure that wiring mess? An FXS channel bank and a BIX strip will save you YEARS in lost time due to wiring and general messiness! Hello Andrew! I'm sorry to ask this really, reeally newbie thing, but... what would be an FXS channel bank, and where would I find more info about some popular models? And the same question goes to... BIX strips! What are those?? :) Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 13:58:58 -0400, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Sunday 15 August 2004 13:50, Francis Augusto Medeiros wrote: I'm sorry to ask this really, reeally newbie thing, but... what would be an FXS channel bank, and where would I find more info about some popular models? And the same question goes to... BIX strips! What are those?? :) haha A channel bank just serves as a device to convert channelized T1s into phone lines and back. FXS ports let you plug phones into them, FXO ports you plug phone lines in to. A BIX strip is just a common wiring closet item that lets you terminate 25 pairs to a strip about 5 or 6 inches long. Google Image search will show you exactly what they look like. A channel bank would typically terminate to a D50F connection, which you would then use a D50M to BIX cable -- this lets you easily terminate 24 phones to a T1, VASTLY reducing the mess and wiring hassle. Since you're moving in to a new place and you don't have existing wiring to make use of I'm not sure any of this would be of any use to you. Thanks Andrew! Well, those options are kinda way above my needs, as we won't have that many phone lines nor extensions. Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help - is voip good for in-house calls?
Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with traditional phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. I've read lots about voip, and I'm quite impressed with it, but most case studies show voip being used to interconnect offices. My case is different - I want to replace a traditional PBX to handle in-house phone calls, either from room-to-room in the same building and room-to-POTS. Any comment, help, tip or link would be greatly appreciated! Yours truly, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Dear Greg, Thanks a lot for your e-mail! Here are my comments: On Sat, 14 Aug 2004 14:37:08 -0700, Greg Broiles [EMAIL PROTECTED] wrote: Asterisk should work fine for this application - but you and/or your users may be expecting the Grandstreams to look/act like traditional key system phones, where you've got a bunch of buttons labeled Computer Room or Joe and Bob, or whatever, where you can press that button to call that extension. The Budgetones don't do that - users will need to remember (or have an extension list to tell them) that the Computer Room is at extension 110, and Joe is at 111, and Bob is 112, and so forth. This is no problem, as with the old PBX we also had to do that, and without LCD. We used a small PBX with regular phones. My concern was if I'd have to teach folks how to dial, but I guess that I can still have the option to assign a number that will give immediate access to the PSTN, so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) My suggestion is that you buy 2 Budgetones and set up Asterisk on an old PC - so your total investment in the experiment will be $200. Get that up running, and let users play with the phones and the functions you can provide. If they like it, great. If they don't like it, you're not out much money, and you ought to be able to resell the Budgetones for something like 80% of the new price on Ebay or whatever. You can get set up with incoming and outgoing IAX connections via someone like Voicepulse or Nufone or IPKall (or some combination thereof) so you can even let people experiment with incoming and outgoing call quality and behavior without spending a lot on interface cards. This is really a great idea. See, my biggest concern is not the voice quality in terms of audio, but if the conversation is allowed to flow in the same way as with regular phones. Or, in other words, if there are significant delays that makes the communication a bit frustrating. Brazilians do interrupt each other a lot while talking on the phone and, on voip calls over the internet with slow connections, this habit made the conversation a bit weird, as sometimes we couldn't realize if the other part started to talk again... :) We don't have in Brazil, as far as I know, voice providers such as those you suggested. So I'll definetly need an FXO. I'm also considering a media gateway - I don't know really if they work the same way for the end user. You might also look at some of the other VoIP phones, which aren't a whole lot more money and might look more like the PBX/key phones that people are used to. The Budgetones are more similar to consumer/home telephones from the early 1990's. I haven't find any other phone below $80. Budgettones, if they work good, seems to be the best option for us. Again, thanks a lot for helping!! Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hi there Wiley! On Sat, 14 Aug 2004 14:43:05 -0700, Wiley E. Siler [EMAIL PROTECTED] wrote: My office build is the same as yours. 15 or so extensions, low traffic 100MB network, and a desire to have a phone system that uses VoIP. I have my system working as a PBX just like you would. I use two TDM400s for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also tested with the Grandstream phones you suggested. SO, we have the same system requirements so here are the answers as I have found them for my implementation Thanks for your e-mail!!! Your setup and your envoironment are really encouraging, since they are very similar to what I have in mind (except for the quantity of POTS lines - we won't use that many). Voice quality on the SIP based phones has a lot to do with the codec you use. The lowest compression codec is uLaw and that is what I use since we have tons of bandwidth to spare. Also, my HP switch has COS (class of service which is like QOS) so I can prioritize the packets coming from my phones over the standard network traffic. Even without this switching feature turned on, performance was great. The phones themselves play another role in the quality. Grandstreams are pretty good and I have only used mine for testing so I will not disparage them. However, the old saying stands. You get what you pay for. Raising your phone budget from $85 to more like $150-250 will get you a phone with more features and greater expandability in my IHO. However, you can still do great things with the cheaper Grandstream phones and still have a system that works very well. IT is all up to what you can spend and what you need. Google the archive by putting site:lists.digium.com in front of your search string (no quotes though). You should see some good info on phones. Well, I'm not really looking for a lot of phone features, just the basics (transfers, call retrieval, etc.). And voice quality is not what I am most worried about, but the delays on the conversation. However, on your mail, you say that latency is, in most cases, unnoticeable, and those are great news to me, as I feel more comfortable to suggest our office to buy ip-phones and use them, knowing they will serve us well. Latency is gonna be there on any network. However, on my network (which is just like yours) the latency is very very low. We are talking 20-40ms tops and it is completely unnoticeable when using the phone. The only problem I have had at all has been with occasional echo. It does not happen often and it usually takes about 5 seconds for the * box to train up and remove it. Most of this seems to originate in the fact that I am using POTS lines. The solution that uses a T1 PRI has better features and I think it has less echo potential. However, that would not work for me since my T1 provider wanted to make me pay 6 grand to switch to a PRI from my standard data T1 with POTS. Just some food for thought... I'll most likely use a BRI. Do you think this will help to avoid echo? I have been a VoIP user for about 1 month after spending another researching what when where how... So, we know I am not an expert... but as a fellow user and new VoIP initiate, I can tell you that Asterisk is a phenomenal product for SMB level offices like yours and mine. When I compared it to a PBX system of comparable power, expandability, and feature set, Asterisk won easily since the only real cost I have had was for my phones. I have my system in place for around 3000 dollars and it is competitive with all the 10K dollar solutions the vendors threw at me plus it has an undeniable advantage in upgrade path. All upgrades to the system are free and the sky is the limit to what you can build using the framework that all the * gurus have built into this system. Not to mention the fact that if anything ever goes wrong with the server, I can have a new one in place in under and hour. Try that with a PBX when some proprietary part goes belly up. You could wait days potentially. My $.02. Hope this helps. That's also what I hope it will happen here! If we want to expand, we don't want to end up with a closed-system that won't handle more extensions or phone lines. And since things are converging, and things like FWD, Vonage and others are helping ppl to communicate, the use of a voip based system would certainly help us more to communicate with our clients and with ourselves. Yours, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 00:22:42 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: My concern was if I'd have to teach folks how to dial, but I guess that I can still have the option to assign a number that will give immediate access to the PSTN, so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) With overlap dialing enabled in Asterisk this should work. We have a similar setup. Note that Asterisk is mostly tested with enbloc-dialing which seems to be the norm in USA where the numbering plan has a fixed length. We have gotten overlap dialing to work correctly except for the call records but they are not that important to us. I hope to create a patch for that after 1.0 has been released. Peter Hej Peter, Hmmm... I'll have to read more about overlap dialing - haven't noticed it while reading the docs. Thanks, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
On Sun, 15 Aug 2004 00:29:21 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote: I'll most likely use a BRI. Do you think this will help to avoid echo? Using a BRI will eliminate echos from the pstn connection. Your ip phones should prevent echos from the local phone connections as well. That way you should not cause any noticable echo for the remote party. Being all digital has its advantages. :-) GREAT news! :)) My uncle's has an old Teles.ISDN card hanging up useless in his computer... :) Cheers, Francis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
