Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. How would / could? one configure * at the remote location to communicate with * at the call center? How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? I know you're all banging your heads on your desks saying "OY! another newbie....". Thanks in advance for your wisdom and guidance. John
Hi John, I'm also new to *, but if you want to set up a callcenter, with 40 people calling the same number at the same time, you probalbly will need a T-1 or E1 line wich AFAIK handles at least 30-calls. You then need at least one Digium E1/T1 card to get the calls into * and other cards to direkt them from * to the phones. I'm researching at this time on what is possible using VoDSL, but I don't dare to say that this might be an alternative for I don't know how many calls can be handled at the same time. But it would be a lot more cost effective than a E1-line here in Belgium. Greetings, Sascha By the way, me Oy! too Am Fr, den 10.09.2004 schrieb John Stegenga um 14:38:> Hi everyone. > I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. > I'm also brand new to *. > I've been reading the Voip.org wiki, and perusing the list archives for a > while since I've been asked to investigate using IP telephone / soft phones > for a call-center type scenario. People (marketing folks) have pointed me > at Cisco, but I really don't wanna. I'd rather be the hero and pull this > off with a much smaller budget. > > Here is a scenario - 40 person call center, all with PC's (windows) and > soft-phone. > -any recommendations on hardware to run *? soft phones? 90% of calls would > be IP / IAX coming to the center. > > I read in the list archives about an ACD application / extension to * that > would probably to what I need in that regard. > - thoughts? > > In remote locations I would also run *, and hook it up to an extension on an > existing PBX. Excuse the complete newbie question, but how many 'wires' do > I need to bring between the PBX and the * box to support multiple > simultaneous calls? These calls would come from any extension on the TDM > pbx to asterisk to the call center. In a typical scenario there would NOT > be a lot of simultaneous calls unless the system we're supporting went down > hard. > > How would / could? one configure * at the remote location to communicate > with * at the call center? > > How would / could? one configure * at the remote location to use the > existing TDM PBX as failover to call the support center via 1-800 if the IP > circuit died? > > I know you're all banging your heads on your desks saying "OY! another > newbie....". > > Thanks in advance for your wisdom and guidance. > > John > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
[sarcasm on] Thank you ALL for your warm welcome to this list. I posted this message yesterday, and since I'm only getting Digest I figured I'd see a response in a day... [sarcasm off] C'mon. This is the Asterisk Users mail list, isn't it? This is where the Voip WIKI tells me to go for information on how people are using *. Even if you only point me in the direction of some other information, it would be great if I could hear SOMETHING from you guys and gals out there....I humbly seek YOUR wisdom. Reposted message: Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. How would / could? one configure * at the remote location to communicate with * at the call center? How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? I know you're all banging your heads on your desks saying "OY! another newbie....". Thanks in advance for your wisdom and guidance. John
----- Original Message -----> > [sarcasm on] > Thank you ALL for your warm welcome to this list. I posted this message > yesterday, and since I'm only getting Digest I figured I'd see a responsein> a day... > [sarcasm off] > > C'mon. This is the Asterisk Users mail list, isn't it? This is where the > Voip WIKI tells me to go for information on how people are using *. Evenif> you only point me in the direction of some other information, it would be > great if I could hear SOMETHING from you guys and gals out there....Ihumbly> seek YOUR wisdom.most of us have spent years/months learning our trade. having new guys who want to "...be the hero and pull this off with a much smaller budget." come in and demand instantaneous help doesn't really fly with this list or any other OS list in my experience. the expectation of the users on this list is to have you TRY something before you come crying for help. you have to remember that responding to anything on this list is totally voluntary. posting things like "sarcasm on" like you have here is pretty much assured you of being ignored other than being flamed. however like S.L.Jackson in Pulp Fiction, you have caught me in a transitional period and I'm not gonna kill you.> > Reposted message: > > snip> > Here is a scenario - 40 person call center, all with PC's (windows) and > soft-phone. > -any recommendations on hardware to run *? soft phones? 90% of callswould> be IP / IAX coming to the center.Call center yes. built into * look on the wiki for stuff about queues and agents. softphones are a bad idea. laptops/desktops are busy running other applications and if your users/computers are like most then they are already overworked with too little horsepower. look into cisco for phones only. they work quite well with * and there are lots available on the open market.> > I read in the list archives about an ACD application / extension to * that > would probably to what I need in that regard. > - thoughts?call center apps are always critical and if you are most interested in the distribution of calls then the built in ACD will work just fine. however, know that the MIS portion of the call center (reports, call vectoring etc) will land squarely on your shoulders to develop something that suits your needs.> > In remote locations I would also run *, and hook it up to an extension onan> existing PBX. Excuse the complete newbie question, but how many 'wires'do> I need to bring between the PBX and the * box to support multiple > simultaneous calls? These calls would come from any extension on the TDM > pbx to asterisk to the call center. In a typical scenario there would NOT > be a lot of simultaneous calls unless the system we're supporting wentdown> hard.FXO/FXS integration would work for very low volume of calls ((qty not sound of course)each channel requires 2 wires to be connected to a port of opposite signalling on your system)) if you are really looking for more seamless integration then look to T-1/PRI networking and I wont get into that here.> > How would / could? one configure * at the remote location to communicate > with * at the call center?IAX trunks. the wiki can explain further> > How would / could? one configure * at the remote location to use the > existing TDM PBX as failover to call the support center via 1-800 if theIP> circuit died?this is the most difficult question you have asked and I am not sure how to answer this with an * system but I am sure it can be done. with normal pbx you would set primary and overflow routes for particular dialing patterns. probably has something to do with GoToIf cmd. Good luck, and remember the most important thing is to TRY something on your own before you come out to the list and ask how to do it. if you dont want to or cant do this and still want to explore * as an option, post a request to the asterisk-biz list, you are bound to get responses from lots of people who have spent the time to learn how to delploy * as a solution. or look deeper into the wiki for a consultant(s) to do all of it for you. will still probably be cheaper than a traditional solution and that still makes you "...the hero and pull this off with a much smaller budget." Jason Kawakami www.optellabs.com
Hi List, I am trying to setup a box which does the following: - Let incoming calls and outgoing local calls through PSTN. - Let outgoing long distance calls through VoIP. Has anybody done anything similar? Cheers, Jean-Michel.
Benjamin on Asterisk Mailing Lists
2004-Sep-28 01:05 UTC
[Asterisk-Users] Asterisk newbie questions
On Tue, 28 Sep 2004 09:51:57 +0500, Jean-Michel Hiver <hiver.j@wanadoo.fr> wrote:> I am trying to setup a box which does the following: > > - Let incoming calls and outgoing local calls through PSTN. > - Let outgoing long distance calls through VoIP. > > Has anybody done anything similar?Yes, most Asterisk installations will be configured similarly. You have two choices ... 1) Do some reading and learn, then when you have specific questions, ask again here. http://www.voip-info.org/wiki-Asterisk 2) Get yourself a Mac (vintage junk iMac will be good enough), download the Asterisk installation package for MacOSX and the Asterisk Assistants for MacOSX, then use those tools to get started quick and easy without prior learning. http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.
Hello all, I`ve got an AVM c2 card instaled on my SuSE box. I?m having problems configuring its channels. I don?t know how to set up asterisk to use the CAPI channels. I don?t know how to call them. My capi.conf is as follow, [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;#########config de la primera interface CAPI##########333 msn=916733232 incomingmsn=* controller=1 softdtmf=1 accountcodecontext=salientes-rdsi ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 ;###############Config de la segunda interface CAPI################33 msn=916733232 incomingmsn=* controller=2 softdtmf=1 accountcodecontext=salientes-rdsi ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 I just create a context in extension.conf called [salientes-rdsi], but i need to know how tell asterisk to use an CAPI channel. Thanks. Regards from Madrid. Ismael GIl. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/bef6a90b/attachment.htm
On Tue, 2004-09-28 at 10:39 +0200, igil@itranser.com wrote:> > Hello all, > > I`ve got an AVM c2 card instaled on my SuSE box. > I?m having problems configuring its channels. > I don?t know how to set up asterisk to use the CAPI channels. I don?t > know how to call them.The README in the capi source could be a good place to start :) -- Dave Cotton <dcotton@linuxautrement.com>
Benjamin on Asterisk Mailing Lists wrote:>On Tue, 28 Sep 2004 09:51:57 +0500, Jean-Michel Hiver ><hiver.j@wanadoo.fr> wrote: > > >>I am trying to setup a box which does the following: >> >>- Let incoming calls and outgoing local calls through PSTN. >>- Let outgoing long distance calls through VoIP. >> >>Has anybody done anything similar? >> >> > >Yes, most Asterisk installations will be configured similarly. > >You have two choices ... > >1) Do some reading and learn, then when you have specific questions, >ask again here. > >http://www.voip-info.org/wiki-Asterisk > >Thanks for the pointer, it looks great. Cheers, Jean-Michel.
Asterisk is very configurable and should be able to do what you want
just fine.
You would just need a dial plan that was configured properly and a VoIP
provider for your LD.
Extensions.conf is what you want to research on the wiki.
http://voip-info.org/wiki-Asterisk+config+extensions.conf
http://voip-info.org/wiki-Asterisk+dial+plan+-+working+example
You would do something like this... Create a trunk for the PSTN and a
trunk for the IAX
Create a section of your dial plan that matches local numbers...
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9XXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9XXXXXXX,2,Congestion
exten => _9480NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9480NXXXXXX,2,Congestion
Create a section that matches you long distance. Mine is PSTN so it
looks like this...
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
Yours would use the IAX methods (to replace my standard PSTN reference)
shown in the second link I listed above...
Reference these as includes for your incoming contaxt and your dialing
will work as you requested.
More info can also be found by searching using google and
site:lists.digium.com in the search box.
Cheers,
Wiley
-----Original Message-----
From: Jean-Michel Hiver [mailto:hiver.j@wanadoo.fr]
Sent: Monday, September 27, 2004 9:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk newbie questions
Hi List,
I am trying to setup a box which does the following:
- Let incoming calls and outgoing local calls through PSTN.
- Let outgoing long distance calls through VoIP.
Has anybody done anything similar?
Cheers,
Jean-Michel.
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Well, how about that for some solidarity... ;) Uh, your still on the list, mon ami.... W _____ From: Mamadou Lamine KA [mailto:lamineka@chaka.sn] Sent: Tuesday, September 28, 2004 12:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk newbie questions Salut Jean-Michel, Est ce que tu peux etre plus pr?cis sur ce que tu veux faire. I feel free to contact you out of the list because c'est quand meme mieux en francais. Lamine The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/0fe42be9/attachment.htm
Probably just about everyone on the list has done something like this at one time or another. I can't give complete Details, but please look at the following resources: * http://www.voip-info.org/wiki-Asterisk - Asterisk Encyclopedia * http://www.asteriskdocs.org - The Asterisk Docs project has nearly completed volume one which is a detail intro on how to use asterisk. * #asterisk on IRC - Major communication link with fellow asterholics. Good luck, Robert Jackson> -----Original Message----- > From: Jean-Michel Hiver [mailto:hiver.j@wanadoo.fr] > Sent: Tuesday, September 28, 2004 12:52 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Asterisk newbie questions > > > Hi List, > > I am trying to setup a box which does the following: > > - Let incoming calls and outgoing local calls through PSTN. > - Let outgoing long distance calls through VoIP. > > Has anybody done anything similar? > > Cheers, > Jean-Michel. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/as> terisk-users > To > UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users
John Howard
2004-Sep-28 07:09 UTC
[Asterisk-Users] Using Grandstream phone , got this error in *
If a routing device has an interface residing in 2 different subnets then packets travelling though it don?t have to be natted because the connection is routed. It would only be natted if IP masquerading was used or snat/dnat implemented on the traversal of the router. jd -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Thomas Gallaway Sent: 28 September 2004 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Using Grandstream phone , got this error in * david winter wrote:> Ok i am using a grandstream on a 192.168.0.x network, connecting to * > on 65.105.x.x. Which might make you think i am using NAT, but im not > there is a cisco 3600 directly connecting these two networks. so its > not natting between them.Correct me if I am wrong but everytime you translate from an private IP to a public IP that is considered NAT? -- Thomas _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been scanned for viruses by MailController - www.MailController.altohiway.com --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.769 / Virus Database: 516 - Release Date: 24/09/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.769 / Virus Database: 516 - Release Date: 24/09/2004
david winter
2004-Sep-28 07:29 UTC
[Asterisk-Users] Using Grandstream phone , got this error in *
you are right, but that isnt my network setup. cisco has 3 eth(0,1,2) eth0 is a /32 (2 usable IP) providing access to my upstream provider this interface is outside NAT eth1 is a /24 (class c) 192.168.0.x where the sip phone is, this is nat inside eth2 is a /28 65.105.x.x (16 usable IP) this is neither inside nor outside. wheny you connect from 192.168.0.109 to 65.105.x.x i dont see that connection as coming from the IP of the NAT outside interface, i see the 912.168.0.x address directly. if the 192.168.0.109 phone was attmpted to connect to anything not on the 192 or on the 66.x.x.x/28, then i would be NATing. David Winter Senior Network Engineer Planet-Telecom, Inc. Tampa FL (813)901-5182 Office (813)864-3162 Direct (813)817-4204 Mobile (813)881-9762 Fax ------------------------------------------ AIM: mobofool ICQ: 3563403 MSN: dwinter@vt.edu Y!: vt_fool Thomas Gallaway wrote:> david winter wrote: > >> Ok i am using a grandstream on a 192.168.0.x network, connecting to * >> on 65.105.x.x. Which might make you think i am using NAT, but im not >> there is a cisco 3600 directly connecting these two networks. so its >> not natting between them. > > > Correct me if I am wrong but everytime you translate from an private IP > to a public IP that is considered NAT? > > -- Thomas > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Salut Jean-Michel, Est ce que tu peux etre plus pr?cis sur ce que tu veux faire. I feel free to contact you out of the list because c?est quand meme mieux en francais. Lamine -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/4514f129/attachment.htm