Re: [Asterisk-Users] LiveVoip is Bankrupt
are you guys still looking for space? I can donate you some space tell me how much you need? I own a communications company based in Tulsa, OK. OCOSA Communications, LLChttp://www.ocosa.com We don't generally do we just started and mySQL as well give me a quote and I 'll will get you hooked up if your interested! Otis Surratt Jr. Peter Corlett wrote: Matt Riddell <[EMAIL PROTECTED]> wrote: [...] Looks like the Daily Asterisk News will need a new host. So, unless anyone can donate space for a custom php and mysql based site, it will be hosted in either New Zealand or Italy. Offers? How much bandwidth does it consume? ___ 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
Re: [Asterisk-Users] Colocation/Telehousing
I would be glad to have your servers in my Data Center in Tulsa, OK. OCOSA Communications, LLC http://www.ocosa.com Ariel Batista wrote: Sahil Gupta wrote: Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. I would suggest www.race.com Regards, Sahil Gupta VoiceValley ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface?
try this sir, Polycom SpIp- Original Message - From: Garrett Nelson To: Sent: Wed, 30 Mar 2005 10:01:05 -0600 Subject: RE: [Asterisk-Users] username/password for PolyCom IP500 webinterface? I did find that in the admin guide, and it does not work. I have triedPolycom both capitalized and not capitalized. -Garrett Polycom/456Caps are important.Sean___ Asterisk-Users mailing listAsterisk-Users@ lists.digium.comhttp://l ists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman /listinfo/asterisk-users ___ 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
[Asterisk-Users] AMP not working in GUI
I have recently installed Asterisk @ 8.0 and loading it fine and setup the ip addressing and change the default password. But when I access the gui from a computer on the network I can pull up the gui but the amp link doesn't work. http://192.168.1.x/admin doesn't work any leads on this. Regards, --- Otis Surratt Jr. / [EMAIL PROTECTED] --- ___ 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
[Asterisk-Users] Asterisk @ Home 0.8 Question
Hello All, This doesn't seem to work for me! I have installed many times on a system with a 20gig 650 mhz, and 256mb of ram. Via Cyrix Chip Asterisk doesn't seem to be running according to the gui! and the panel doesn't work says Error loading configuration file variables.txt?aldope=98937 What does that mean? Thanks - Otis / [EMAIL PROTECTED] - ___ 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
[Asterisk-Users] Need help getting zap trunk to work
Hello All, I am currently using Asterisk @ Home 9.0 and I have the Digium Wildcard X100 / X100 I purchase off ebay, a old PIII and nic. I can get the trunk to show up in the flash operator panel thing or be active I dont know what it is but it doesn't show up once it is add and created and i even setup the out going plans etc. Regards, Otis Surratt Jr. ___ 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
Re: [Asterisk-Users] Need help getting zap trunk to work
ok sir thanks Mike Price wrote: [EMAIL PROTECTED] hove now released version 1.0. According to the changelog this is supposed to fix the ZAP problem. I am trying it myself today, for the same reason. I'll post my result. Mike On Tue, 2005-05-03 at 06:35, Listacc wrote: Hello All, I am currently using Asterisk @ Home 9.0 and I have the Digium Wildcard X100 / X100 I purchase off ebay, a old PIII and nic. I can get the trunk to show up in the flash operator panel thing or be active I dont know what it is but it doesn't show up once it is add and created and i even setup the out going plans etc. Regards, Otis Surratt Jr. ___ 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 ___ 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 ___ 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
[Asterisk-Users] does asterisk work with other processors
Hello All, I have tried numerous versions of asterisk from asterisk at home to compiling it myself through the cvs server. I don't understand it works fine with the intel p2 box but not the faster via cyrix box. Is it the processor or something? Regards, Otis Surratt Jr. / [EMAIL PROTECTED] ___ 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
[asterisk-users] Transfer Call to Cell Phone
Hello All, I apologize if this question has already been answered but how do you transfer a call to a cell phone or another land line outside the PBX? Setup I have two pots lines into my current Asterisk Box. I have an outsides sales guy who wants to work off his cell phone or transfer his calls from his extension and the main sales extensions. How can I do this right? -- Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Call to Cell Phone
Well seems like I am already doing first method minus the extension. We do have full features on our lines so both lines are free once the transfer is complete. We also have toll calls on our lines so it would not be a problem, so I do not have to worry about AT&T dropping the call. I tried to create a separate extension for this but it did not work. The only way I can get it to work is by have the call on the 1st line then transfer it out on the 2nd line. After that is complete both lines are free. A worst case scenario would be where our sales volume picked and we needed to transfer a call and could not because of the slots are filled. Can you give an example of creating an extension which points to a cell phone. Secondly how can you have if no one answers an extension it dials the cell number next. That maybe answered in the example. I have the system setup so it just dials out which ever line is not busy. Thanks! Otis John Faubion wrote: >> Polycom Phones >> 1. New call >> 2. Press 9 access outside line >> 3. Dial Cell Number >> 4. Transfer the call that way. >> > > Once you initiate a new call you will tie up the second line. Your asterisk > box will now be bridging the two lines. The lines will stay tied up until > the salesman drops the call. > > One method you might be able to employ here would be to add a call transfer > to the pots lines. Then you would need to send a hook flash to the pots > line, and dial the salesman's number when you get the dial tone. Then, > depending on how your local Telco supports the call transfer feature, you > may be able to free up the line. Not all Telcos support this the same way as > some consider it a method of toll avoidance and thus drop the call. This > would be possible in an area where a call from party A to party B is a local > call and the call from party B to party C is a local call but a call from > party A to party C would be a toll call. Since the call from party A to > party C is a toll call, the Telco may opt to drop the call. If the transfer > part works, there may even be a way to setup the dial plan to intercept your > phones call transfer feature and use a 1-2 digit code to select which phone > number to send out. I have not done this but I think it is reasonable as > I've heard of home users doing it. > > By the second option, are you talking about the TDMA/GSM gateway? If so, yes > this is pretty slick. We considered it initially as well. Our decision not > to use it was based on the fact that many of our agents are on different > mobile plans. I think when we requested the info from the agents we had 6 > different wireless companies represented. Since Sprint/Nextel, > Cingular/AT&T, T-Mobile don't like to play nicely with other, we didn't > expect to see any real savings from the free mobile to mobile calls. This > was mainly due to the fact that we don't pay for the agents phones and thus > we can't really tell the agents which carriers to use. I do know of a couple > of installations where the company does provide the phones and I understand > the savings can be significant. I was told by friend that the box they > installed paid for itself in just a couple of months. But their phone were > already on the same plan. > > John > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Call to Cell Phone
John Faubion wrote:
>> We do have full features on our lines so both lines are free once the
>> transfer is complete. We also have toll calls on our lines so it would
>> not be a problem, so I do not have to worry about AT&T dropping the
>>
>
> The issue really isn't whether you have the ability to make toll calls on
> your line. The concern here is in what the regulatory agencies call "toll
> bridging" which is using a system to relay a call from one local calling are
> to another local calling area to avoid a toll charge. This is one of those
> gray areas that can become a problem if your not careful. The problem comes
> up if you have customers that can call you as a local call and you are
> forwarding them on to another party that is a local call for you but would
> be a toll call for the customer. This is essentially what toll bridging is
> about. Now your not likely to have to worry about the legal ramifications of
> this since your merely connecting the customer with an extension of your
> company, namely your salesman. Where this could become a problem for you
> would be in transferring the customer using the same pots line. The reason
> is that AT&T is handling the transfer. When you transfer the call, it
> essentially becomes a new call. The main difference is that you have
> provided the called number. So the software in the Class 5 (End office)
> switch, takes the number you provide and runs the call through its routing
> translations (similar to the Asterisk dialing plan) and if it determines
> that the destination number is outside the originators Local Area Transport
> Area or LATA, then it will either drop the originator to a message that
> says, "You must first dial a 0 or 1 before calling this number" or it may
> deny the transfer allowing you to stay connected to the customer. Neither
> one looks very professional. The only way around this would be to provide
> another line or trunk to pass the call down. Now if your not in an
> overlapping LATA this probably isn't an issue.
>
> John you a right about the LATA I know I am in one LATA 536 or 538 for
> eastern OK. But I do not know the LATA on the Wireless which is now AT&T. So
> I will keep a watch out for it. Thanks for the tip!
>
>> The only way I can get it to work is by have the call on the 1st
>> line then transfer it out on the 2nd line. After that is complete both
>> lines are free.
>>
>
> Are you saying that you are able to route a call from line 1 to line 2 and
> have the call transfer, thus freeing the lines or that once the call
> completes the lines are freed? I've never seen the first scenario. The
> second scenario is the normal behavior.
>
> I am saying here that I can transfer the call from line 1 to line 2 and once
> I transfer off the asterisk box it frees the two phone lines. My whole
> arguement was to find a solution for doing this automatically on the basises
> of dial an extension which can just transfer it to the cell phone. So ext
> 4001 cell-1 ext 4002 cell-2 etc. I do not mid doing it manually. But thanks
> for the help!
>
>> Can you give an example of creating an extension which points to a cell
>> phone. Secondly how can you have if no one answers an extension it dials
>> the cell number next. That maybe answered in the example.
>>
>
> In extensions.conf use something like this.
> [global]
> SIP-PROV = "sip.urprovider.com"
> ; Now set the call forward numbers
> CFN21 => "551234" ; These are normally set in an external file
>
> [internal]
> exten => 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})
>
> [macro-stdext];
> ; Standard extension macro:
> ; ${ARG1} - Device(s) to ring
> ; ${ARG2} - Our call forward number
> exten => s,1,Dial(${ARG1},10)
> exten => s,2,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}>0]?s-CFWD,1)
> exten => s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
> exten => s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
> exten => s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
> exten => s-CFWD,2,Goto(s-NOANSWER,2)
> exten => _s-.,1,Goto(s-NOANSWER,2)
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
> There is more to this but this should show the basics of what we use. I
> store my Call Forward Numbers (CFN) in an external file. This allow me to
> update the file externally (currently with a web interface but as soon as I
> get the prompts recorded it will be done with an IVR) and then just reload
> the extensions to activate the new numbers. Also I using SIP for pretty much
> everything. Our TDM400 doesn't even have modules, it's just there for
> timing. However you should be able to convert the SIP calls to ZAP calls for
> you use. The internal context is included in our default context. Dialing
> extension 21 calls the stdext macro. This dials the local extension first.
> If not answered after 10 seconds, we check to make sure we have a phone
> number to send the call out with. If not we send it on to voice mail.
> Otherwise we send it to the s-CFWD. The ch
Re: [asterisk-users] Transfer Call to Cell Phone
Ryan Goldberg wrote:
> OCOSA ListAcc wrote:
>
>> Can you give an example of creating an extension which points to a cell
>> phone. Secondly how can you have if no one answers an extension it dials
>> the cell number next. That maybe answered in the example. I have the
>> system setup so it just dials out which ever line is not busy. Thanks!
>>
>
> I'm quite new to *, but I've got this in place in my first rendition, and
> I'm pretty sure it does what you want:
>
> exten => 101,1,Dial(SIP/${EXTEN},15,t)
> exten => 101,n,Dial(Zap/4/12185551212,30,tpm)
> exten => 101,n,VoiceMail([EMAIL PROTECTED])
> exten => 101,n,Playback(vm-goodbye)
> exten => 101,n,Hangup
>
> caller dials extension 101. It first tries his desk for 15 seconds, then
> it tries his cell over a zap channel (the 'p' turns on call screening),
> then it finally hits voicemail. In our actual dialplan, the cell phone
> call goes out over sip, so the line looks like this:
>
> exten => 101,1,Dial(SIP/lesnet/12185551212,30,tpm)
>
> Alternatively, the first line could be:
>
> exten => 101,1,Dial(SIP/${EXTEN}&Zap/4/12185551212,30,tpm)
>
> which would dial both the desk and the cell at the same time...
>
> See http://www.voip-info.org/wiki-Asterisk+cmd+Dial
>
> Hope that helps.
>
> Ryan
>
Great I will look it over this weekend and see if it works!!! Thanks!
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[asterisk-users] Asterisk 1.2.16 - No Caller ID
Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 -- Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp ___ --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 1.2.16 - No Caller ID
Eric, I have watched the CLI before and it said nothing although I did change the position of the callerid = asreceived to right below and nothing it still shows up on the phones "asterisk" and in voice mail sent via e-mail unknown caller: Here is an output from a while back but it stopped so I do not know what is up. -- Starting simple switch on 'Zap/1-1' Apr 6 16:54:39 ERROR[4726]: callerid.c:276 callerid_feed: fsk_serie made mylen < 0 (-92) Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6255 ss_thread: CallerID feed failed: Success Apr 6 16:54:39 WARNING[4726]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/1-1' Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Eric "ManxPower" Wieling wrote: I don't know where he got the bizarre "useincomingcalleridonzaptransfer" option, but it does not exist as you can see below: [EMAIL PROTECTED] ~]# grep useincomingcalleridonzaptransfer /home/software/asterisk/asterisk-1.2.17/configs/zapata.conf.sample [EMAIL PROTECTED] ~]# Maybe the option is specific to BRIstuff patches to Zaptel. You want the following before your FXO ports in /etc/asterisk/zapata.conf: usecallerid=yes callerid=asreceived You will also want to watch the console when a call comes in to see if there are any Caller*ID errors. OCOSA ListAcct wrote: Giorgio, That does not work it just shows up as useincomingcalleridonzaptransfer I set the following: callerid=useincomingcalleridonzaptransfer. Are you referring to something else like useincomingcalleridonzaptransfer=yes Otis Surratt Jr. / [EMAIL PROTECTED] OCOSA Communications, LLC PH: 918.585.9882 x 205 Fax: 918.585.5857 Visit Tulsa's Best Internet Data Center: http://www.ocosa.com/hosting/colo/index.asp Giorgio Incantalupo wrote: Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please help!! I have two pots lines coming into the Asterisk Box caller ID is set in the zapta.conf Here is what our zapata.conf looks like [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.5 txgain=5.5 group=1 pickupgroup=1-4 immediate=no context=bell signalling=fxs_ks callerid=asreceived channel=1 channel=2 ___ --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 ___ --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
