Re: [asterisk-users] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
We have written stuff previously for most major phones that does
auto-deploymentserver sits there waiting for phone to ask for
configs, when the phones hit the server, the configs are written on the
fly.
Bit fiddly to write, but once it's going it's pretty good.
PaulH
On Sat, 2007-10-20 at 22:58 -0700, Philip Prindeville wrote:
> Erik Anderson wrote:
> > On 10/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >
> >> If you are trying to use non-complied ("XML") profiles... don't even
> >> bother wasting your time.
> >>
> >
> > Why is that? I'm using the xml-style config and they're working just fine.
> >
> >
>
> I'd like to be able to templatize a server, add a bunch of new handsets
> into sip.conf and extensions.conf, and then plug the phones into a
> network and have some DHCP and/or TFTP "glue" logic that sees the DHCP
> or TFTP request, and from it generates a boot file (an .XML file) and a
> response parameter list for DHCP... populates a file into the /tftpboot/
> directory, etc.
>
> How viable is this?
>
> I'd like it to be lightweight enough that it could be done on some of
> the smaller embedded Asterisk boxes (like the 400MHz SoHo units).
>
> Thanks,
>
> -Philip
>
>
>
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Re: [asterisk-users] Extensions.conf for basic IVR?
On 10/20/07, Vincent <[EMAIL PROTECTED]> wrote: > I've never built an IVR before, so I was wondering if someone > could share some code from their extensions.conf that would perform > some of thoses steps: Try Google for asterisk ivr The first ten sites that come up, including voip-info.org, usually a good place to look first, each have full examples. Look also for the background application wich is used to play the file, get input and jump to the extension entered. ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Per Jessen wrote: > Luki wrote: > >> Here's how you do it. >> [snip] > > Oh well - I wonder what I'm doing wrong then. I've been trying to get > this to work for most of last week. Luki, thanks for writing to say it DOES work. I've have just now had another look, found my mistakes (basically $MAC instead of $MA), and it's working! > Do you push it from Asterisk or somewhereelse? Again, I can't make it > work. I've got "Auth resync-reboot" disabled on the SPA, so it > shouldn't be asking for authentication, but the SIP NOTIFY goes out, > and the phone does nothing. Got that working too. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX
On this machine its the first install, but i get this error 3 month before on an other machine also. I think the debug will bring t much data, cause there is any half second a call try, and its really hard to find this error in the debug file. The only thing i know is if i use a Sangoma card, the problem went. Can i send you a BIG debug file where some of this errors happened? Thanks Nico On Fri, 19 Oct 2007, Matthew Fredrickson wrote: > [EMAIL PROTECTED] wrote: >> Hi, >> >> I'm running some Asterisk-machines, and on one of them i get this errors >> in the CLI, but i don't know what that means. >> >> Hardware: >> Digium 4-Port E1 Card with HWEC >> Intel Pentium D 3 GHz >> 2 GB RAM >> SATA Harddisk >> Supermicro Mainboard >> >> Software: >> latest libpri/zaptel/asterisk of version 1.2 >> >> I tried also asterisk version 1.4.x, but there the problem occurs every 10 >> calls, on asterisk 1.2 its about every 100 calls. > > Did this recently start, like after you upgraded or is this something > that has always been a problem for you since you installed? > > If it has always been a problem, can you post a `pri debug span x` trace > of a call when this happens? That will help to know more about what is > going on here. > > -- > Matthew Fredrickson > Software/Firmware Engineer > Digium, Inc. > > ___ > --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
[asterisk-users] 16 ports wanted
Hi all, I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 or 2 TDM808 to get 16 FXO? What is the difference (in performance and control) in using 4 x TDM404 and 2 x TDM808 if possible? ango ___ --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] 16 ports wanted
Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote: > Hi all, > I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 > or 2 TDM808 to get 16 FXO? What is the difference (in performance and > control) in using 4 x TDM404 and 2 x TDM808 if possible? > ango Well, using more than one TDM card in your PC is not a good idea, because of interrupts. If you have to have 16 FXO you can more options: 1. Using TDM2400P with 4 FXO modules ($1775) 2. Using Xorcom's Astribank (external) ($1170) 3. Using some T1/E1 card with Channel Bank (more expensive) -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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
[asterisk-users] Issue Nortel CS2K/ISN08 to Asterisk Trixbox
I have an trixbox(asterisk) software on a pc home edition. Origination is a Nortel ,model=CS2K,version=ISN08 and my asterisk is doing termination.Nortel sent calls to us ,Asterisk and they said that is sending call and i saw the trace as following: sip: [EMAIL PROTECTED] IP:5060 ;user phone but in my CDR i can view origination number but at destination i get 200 ,they said to me that they sent correct destination number in form of: CC+ area code +telephone number We do not have users or passwords. Nortel is given to me IP Signalling (1 IP) and media IP , the scenario is IP to IP. In my side,asterisk ,i configured: 1. SIP trunk with : maximum channels=2 Outgoing dial rule=cc+area +. Trunk peer == allow=all context=from-internal host=Nortel Signalling IP port=5060 type=peer Incoming setting on SIP trunks=nothing register string=nothing 2.ZAP trunk outgoing dial rules === dial rules:cc+area code+. Please can you helping with this configuration or how i can configure ,the calls to come from Nortel to my asterisk? Nortel CS2k ISN08 .>SIP.>Asterisk trixbox Any help it will be higly appreciated Many thanks in advance, Tiberiu ___ --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
[asterisk-users] Astmanproxy issues
Hello *, I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens that I send a request and I receive a response to ANOTHER request that it got in the frame time between my request and my response. Did anyone else notice this behaviour? How can this be solved? I've been reading the source code, but I didn't find a solution. Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ --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
[asterisk-users] Call Hold
Hello Again, I was just wondering if anyone can give me a heads up regarding the possibility of identifying that a user currently in an active call is also being dialled by another extension. Does asterisk/sip issue an event that says there's a call attempting to reach you? If so, I will then use this information to update an interface which will then give the user the opportunity to put the current line on hold and take the waiting call. Is it possible to put a user on hold and take another call? Regards James This email and any attachments are confidential to the intended recipient and may also be privileged. If you are not the intended recipient please delete it from your system and notify the sender. You should not copy it or use it for any purpose nor disclose or distribute its contents to any other person. ___ --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] Astmanproxy issues
Ciao Andrea, > I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens > that I send a request and I receive a response to ANOTHER request that it got > in the frame time between my request and my response. > > Did anyone else notice this behaviour? How can this be solved? > > I've been reading the source code, but I didn't find a solution. I'll answer to myself, in order to help others with the same problem. The 'autofiltering' capability of astmanproxy can avoid this problem. If it's set to 'on', the user will receive only the responses that contain the same ActionID that was specified in the request. HTH, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ --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
[asterisk-users] Snom 360 lights not working on subscription
Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --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] Call Hold
If your SIP phone supports multiple appearances for a line, you should just get another INVITE coming in while you are on your current call. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Lees, James (UK) > Sent: 22 October 2007 13:40 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Call Hold > > > Hello Again, > > I was just wondering if anyone can give me a heads up > regarding the possibility of identifying that a user > currently in an active call is also being dialled by another > extension. > > Does asterisk/sip issue an event that says there's a call > attempting to reach you? If so, I will then use this > information to update an interface which will then give the > user the opportunity to put the current line on hold and take > the waiting call. > > Is it possible to put a user on hold and take another call? > > Regards > > James > > > This email and any attachments are confidential to the > intended recipient and may also be privileged. If you are not > the intended recipient please delete it from your system and > notify the sender. > You should not copy it or use it for any purpose nor disclose > or distribute its contents to any other person. > > > > ___ > --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] Call Hold
Lees, James (UK) wrote: > Hello Again, > > I was just wondering if anyone can give me a heads up regarding the > possibility of identifying that a user currently in an active call is > also being dialled by another extension. > > Does asterisk/sip issue an event that says there's a call attempting to > reach you? If so, I will then use this information to update an > interface which will then give the user the opportunity to put the > current line on hold and take the waiting call. > > Is it possible to put a user on hold and take another call? > I would surmise that some or most of the answer to this is related to your SIP (or whatever you use) client phone. I use a soft sip phone on my Linux desktop computer called Twinkle. This supports two connections so I can toggle between two calls without Asterisk even having to know about it... If your client device only supports one "line" then Asterisk will see it as busy if it tries to route another call to it. So you would have to do something with that information in your dialplan. Whether your client could receive and act upon additional messages depends on it's capabilities. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ --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
[asterisk-users] Video Conference
Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am considering at this point is hacking up my own solution using off the shelf equipment. Decent web camera, Polycom conference phone(maybe if the budget holds) and a large wide screen LCD monitor all connected to * Sound reasonable or am I living a pipe dream? JohnM ___ --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] Video Conference
Direct single line video conferencing via SIP is actually pretty straightforward and works rather well. Multipoint conferencing is where you get into a bit of a mess. There are precious few products out there that claim multipoint SIP video conferencing capability, and we've had no luck so far with any of it being what one might consider straightforward. N. John Millican wrote: > Hello All, > I am looking at doing some video conferencing with SIP. I was hoping to get > some early pointers from any one that is currently doing this. I have been > all over goggle and voip-info and there is a ton of anecdotal information > but, I was hoping for more specifics of what people are actually using that > works and even some of what hasn't worked so that I can stay away. What I am > considering at this point is hacking up my own solution using off the shelf > equipment. Decent web camera, Polycom conference phone(maybe if the budget > holds) and a large wide screen LCD monitor all connected to * > Sound reasonable or am I living a pipe dream? > JohnM > > > ___ > --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
[asterisk-users] Making Asterisk a "Voice Router"
Hi, I'm interested in what software (Free or course) that people use when they want to add a "dial by voice" service to their asterisk system. Meaning I pick up the phone.. dial some extension. it prompts me for name.. I say "John Smith".. and it dials his extension and connects the call.. TIA, ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote: > The SPA-9x1 does support http download, but I don't see how you could > change the initial TFTP request to HTTP without manually configuring > the phone. Even then I'm not sure it would work - I certainly haven't > managed to make any of my SPAs do an auto-config over HTTP. Actually, it's really easy to do. Here's a copy of my spa942.cfg file which I use to point the phone at my web server, as well as upgrade the firmware. http://192.168.0.103/AutoProvision/prov.php?mac= $MA&dev=spa$PSA 300 192.168.0.103 60 60 (! 5.1.5)?tftp://192.168.0.103/spa942-5-1-5.bin -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Issue Nortel CS2K/ISN08 to Asterisk Trixbox
[EMAIL PROTECTED] wrote: > I have an trixbox(asterisk) software on a pc home edition. > Origination is a Nortel ,model=CS2K,version=ISN08 > and my asterisk is doing termination.Nortel sent calls > to us ,Asterisk and they said that is sending call > and i saw the trace as following: > > sip: [EMAIL PROTECTED] IP:5060 ;user phone > > but in my CDR i can view origination number but > at destination i get 200 ,they said to me that they sent > correct destination number in form of: > > CC+ area code +telephone number > > > We do not have users or passwords. > > Nortel is given to me IP Signalling (1 IP) and media IP , > the scenario is IP to IP. > > In my side,asterisk ,i configured: > > 1. SIP trunk with : > > maximum channels=2 > > Outgoing dial rule=cc+area +. > Trunk peer > == > allow=all > context=from-internal > host=Nortel Signalling IP > port=5060 > type=peer > > Incoming setting on SIP trunks=nothing > register string=nothing > > > > 2.ZAP trunk > >outgoing dial rules >=== >dial rules:cc+area code+. > > > Please can you helping with this configuration or how i can > configure ,the calls to come from Nortel to my asterisk? > > Nortel > CS2k > ISN08 .>SIP.>Asterisk trixbox > > > > Any help it will be higly appreciated > > Many thanks in advance, > > Tiberiu > > > ___ > --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 > Change your type to friend not peer. This may help. Turn debug on in the cli console for the ip address that the calls are coming from, this way you can see if the info that they are sending is correct. Jonn ___ --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] Video Conference
> John Millican wrote: > > Hello All, > > I am looking at doing some video conferencing with SIP. I was hoping to > > get some early pointers from any one that is currently doing this. I > > have been all over goggle and voip-info and there is a ton of anecdotal > > information but, I was hoping for more specifics of what people are > > actually using that works and even some of what hasn't worked so that I > > can stay away. What I am considering at this point is hacking up my own > > solution using off the shelf equipment. Decent web camera, Polycom > > conference phone(maybe if the budget holds) and a large wide screen LCD > > monitor all connected to * > > Sound reasonable or am I living a pipe dream? > > JohnM > > On Monday October 22 2007 9:32 am, SIP wrote: > Direct single line video conferencing via SIP is actually pretty > straightforward and works rather well. > > Multipoint conferencing is where you get into a bit of a mess. There > are precious few products out there that claim multipoint SIP video > conferencing capability, and we've had no luck so far with any of it > being what one might consider straightforward. > > N. Thanks for the responce. Have you had any luck at all even with what one might not consider straight forward? I am trying to avoid paying the $1000+ per location needed to purchase something from say Polycom or Tandberg. I would even be willing to do something along the lines of a web app for video and some how tie that together with the voice through Asterisk. Just don't want to look like one of the old dubbed over Japanese movies from when I was a kid (lips move and then a couple seconds later you hear voice). JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prompting for number when CID number not sent?
On Sun, 2007-10-21 at 17:22 +0200, Vincent wrote:
> ;here, rewrite CID name by looking up CID # in database
> ;put CID name + number in variables
> ;exten => _[1-4],n,SetVar(cid=${callerid})
> ;send e-mail with CID name + number and link to WAV file to people in
> charge of selected software
Instead of ${callerid} here (which probably isn't working for you
anyway), you probably want to use the CALLERID dialplan function to
retrieve the CallerID number, like this:
exten => _[1-4],n,Set(cid=${CALLERID(num)})
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Making Asterisk a "Voice Router"
On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: > I’m interested in what software (Free or course) that people use when > they want to add a “dial by voice” service to their asterisk system. > Meaning I pick up the phone.. dial some extension… it prompts me for > name.. I say “John Smith”.. and it dials his extension and connects > the call.. I've done this using Asterisk and the LumenVox speech engine... in fact, I spoke about it at AstriCon Europe in 2006. My slides are available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be slightly out of date, but it should at least get you started.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Making Asterisk a "Voice Router"
Coool... thanks man.. do you have any installation procedures or notes? Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Monday, October 22, 2007 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Making Asterisk a "Voice Router" On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: > I’m interested in what software (Free or course) that people use when > they want to add a “dial by voice” service to their asterisk system. > Meaning I pick up the phone.. dial some extension… it prompts me for > name.. I say “John Smith”.. and it dials his extension and connects > the call.. I've done this using Asterisk and the LumenVox speech engine... in fact, I spoke about it at AstriCon Europe in 2006. My slides are available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be slightly out of date, but it should at least get you started.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Video Conference
Hi, We have done a video and voice conferencing application but it's still Alpha. We use Red5/Flash for video, IAX for audio. You can take a look at http://code.google.com/p/blindside/ and click on the screencast and Webconference demo. Maybe we can work with each other to further improve it. Richard On 10/22/07, John Millican <[EMAIL PROTECTED]> wrote: > > > > > John Millican wrote: > > > Hello All, > > > I am looking at doing some video conferencing with SIP. I was hoping > to > > > get some early pointers from any one that is currently doing this. I > > > have been all over goggle and voip-info and there is a ton of > anecdotal > > > information but, I was hoping for more specifics of what people are > > > actually using that works and even some of what hasn't worked so that > I > > > can stay away. What I am considering at this point is hacking up my > own > > > solution using off the shelf equipment. Decent web camera, Polycom > > > conference phone(maybe if the budget holds) and a large wide screen > LCD > > > monitor all connected to * > > > Sound reasonable or am I living a pipe dream? > > > JohnM > > > > On Monday October 22 2007 9:32 am, SIP wrote: > > Direct single line video conferencing via SIP is actually pretty > > straightforward and works rather well. > > > > Multipoint conferencing is where you get into a bit of a mess. There > > are precious few products out there that claim multipoint SIP video > > conferencing capability, and we've had no luck so far with any of it > > being what one might consider straightforward. > > > > N. > > Thanks for the responce. Have you had any luck at all even with what one > might not consider straight forward? I am trying to avoid paying the > $1000+ > per location needed to purchase something from say Polycom or Tandberg. I > would even be willing to do something along the lines of a web app for > video > and some how tie that together with the voice through Asterisk. Just > don't > want to look like one of the old dubbed over Japanese movies from when I > was > a kid (lips move and then a couple seconds later you hear voice). > JohnM > > > > > ___ > --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] Making Asterisk a "Voice Router"
Nice job! I took the liberty to post it on AstPligg as well: http://tinyurl.com/268bac Thanks l. In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith <[EMAIL PROTECTED]> ha scritto: > On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: >> I’m interested in what software (Free or course) that people use when >> they want to add a “dial by voice” service to their asterisk system. >> Meaning I pick up the phone.. dial some extension… it prompts me for >> name.. I say “John Smith”.. and it dials his extension and connects >> the call.. > > I've done this using Asterisk and the LumenVox speech engine... in fact, > I spoke about it at AstriCon Europe in 2006. My slides are available at > http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be > slightly out of date, but it should at least get you started.) > > -- Home of QueueMetrics - http://queuemetrics.com ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
> Luki, thanks for writing to say it DOES work. I've have just now had > another look, found my mistakes (basically $MAC instead of $MA), and > it's working! I'm glad you got it sorted out. Yes, it works with XML or compiled files. To help with troubleshooting, specify a syslog server and set the debug level to 3 in the initial spaXXX.cfg, and the device will tell you what it tried, what worked and what failed (i.e. XML parse error, invalid parameter, URLs, etc.). That's just a note should someone get hang up on that in the future. --Luki ___ --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] tech prefix
no that didnt work.
- Original Message -
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X. => {
Dial(tech/123{EXTEN});
}
?
Regards,
Philipp Kempgen
--
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Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
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Re: [asterisk-users] tech prefix
Here's what worked:
exten=>_X.,1,Dial(SIP/"prefix"[EMAIL PROTECTED] trunk)
substitute "prefix" for the tech prefix you would like to append.
-Jon
- Original Message -
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X. => {
Dial(tech/123{EXTEN});
}
?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
Check out again http://spc.pifiu.com it seems the owner of the site
has added the latest admin guide for SPA-900 series & the spc.exe for
5.1.5 & 5.1.7 firmware.
On 10/21/07, Per Jessen <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] wrote:
>
> > If you are trying to use non-complied ("XML") profiles... don't even
> > bother wasting your time.
>
> Oh. I _am_ using the XML format. When I initiate a resync over the
> http server, it works fine, except the SPA doesn't start the regular
> resync.
>
>
>
> /Per Jessen, Zürich
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Re: [asterisk-users] Making Asterisk a "Voice Router"
Is this free? I see the tuner is free.. but the speech rec isn’t? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Monday, October 22, 2007 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Making Asterisk a "Voice Router" Nice job! I took the liberty to post it on AstPligg as well: http://tinyurl.com/268bac Thanks l. In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith <[EMAIL PROTECTED]> ha scritto: > On Mon, 2007-10-22 at 09:39 -0400, end1r wrote: >> I’m interested in what software (Free or course) that people use when >> they want to add a “dial by voice” service to their asterisk system. >> Meaning I pick up the phone.. dial some extension… it prompts me for >> name.. I say “John Smith”.. and it dials his extension and connects >> the call.. > > I've done this using Asterisk and the LumenVox speech engine... in fact, > I spoke about it at AstriCon Europe in 2006. My slides are available at > http://www.astricon.net/files/Jared_Smith_EUR06.pdf. (They may be > slightly out of date, but it should at least get you started.) > > -- Home of QueueMetrics - http://queuemetrics.com ___ --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
[asterisk-users] Split asterisk in two ?? One TDM and One IP only??
I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am thinking of doing. Server1 Keep one physical server just for TDM functions. PRI to Telco PRI to old PBX for Fax. (basically using it as a mux) Keep meetme here for Digium card timing. Server2 Build a new asterisk install within Xen VM with data stored on an iSCSI SAN. This would be all IP. IAX and SIP extensions. IAX and SIP providers. IVR Voicemail Web access to voicemail CDR This way I can test different versions of the features of Server2 (clone with different IP) without affecting production. I assume that I just use an IAX or SIP trunk between the two asterisk servers. Does this make sense? Are others doing similar? Are there any other features that require the TDM card besides PRI, Fax and Meetme? I have heard of people using Xen for IP only asterisk, but are there any known gotchas? Thanks, -- -- Steven http://www.glimasoutheast.org ___ --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
[asterisk-users] Authenticate by IP?
I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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
[asterisk-users] Polycom 601 + Headset
Hi List, I am using a Plantronics CS50 head set with my Polycom 601. I use the button on it to pick up calls. Is there any way to have the phone set up that if I pick up with the button on the headset that it sends the call to the headset and that I don't have to press the headset button on the phone every time ? Thanks. Dovid___ --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] tech prefix
Jon Weisman wrote:
> Here's what worked:
>
> exten=>_X.,1,Dial(SIP/"prefix"[EMAIL PROTECTED] trunk)
>
> substitute "prefix" for the tech prefix you would like to append.
> - Original Message -
> From: "Philipp Kempgen" <[EMAIL PROTECTED]>
> To: "Asterisk Users"
> Sent: Tuesday, October 16, 2007 3:09 PM
> Subject: Re: [asterisk-users] tech prefix
>
>
> Jon Weisman wrote:
>
> How can I add a prefix to an outbound call?
>
> _X. => {
> Dial(tech/123{EXTEN});
> }
That's what I said.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Authenticate by IP?
On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > I have a customer that needs an Asterisk server to sell minutes for > cell phones in Mexico. I do not see a problem with that since he will > get the calls by SIP and then use GSM adapters to get the calls into the > GSM network. My problem is that his customers only want to be > identified by IP and not by a username and password. Is there a way to > authenticate just by using an IP address? > There certainly is. ___ --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
[asterisk-users] [France CID] Does Zaptel support ETSI FSK?
Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium TDM400 clones like those from OpenVox? Thank you. ___ --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] Authenticate by IP?
Saludos Carlos, Como vas a recibir las llamadas via SIP, puedes especificar el IP del host que te enviara las llamadas, por ej. Este es un bloque que tengo definido en el SIP.conf de uno de mis servers para enrutar las llamadas internacionales y a telefonos moviles utilizando un proveedor de terminacion. [oficina] type=peer context=from_office ; Esto va a mi 'extensions.conf' host=200.88.42.29; Este es el ip publico en la oficina (estatico) nat=no canreinvite=no qualify=yes disallow=all allow=g729 allow=ulaw Creo que eso contesta tu pregunta. -- Jose P. Espinal slackware-es.com Carlos Chavez wrote: I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? ___ --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] Prompting for number when CID number not sent?
On Mon, 22 Oct 2007 10:14:41 -0400, Jared Smith <[EMAIL PROTECTED]>
wrote:
>Instead of ${callerid} here (which probably isn't working for you
>anyway), you probably want to use the CALLERID dialplan function to
>retrieve the CallerID number, like this:
Thanks for the tip. It'll come in handy... once I finally get the TDM
card to report CID :-)
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[asterisk-users] dial-out call queue
Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira ___ --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] Authenticate by IP?
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote: > Saludos Carlos, > > Como vas a recibir las llamadas via SIP, puedes especificar el IP del > host que te enviara las llamadas, por ej. > > Este es un bloque que tengo definido en el SIP.conf de uno de mis > servers para enrutar las llamadas internacionales y a telefonos > moviles utilizando un proveedor de terminacion. > > [oficina] > type=peer > context=from_office ; Esto va a mi 'extensions.conf' > host=200.88.42.29; Este es el ip publico en la oficina (estatico) > > nat=no > canreinvite=no > qualify=yes > disallow=all > allow=g729 > allow=ulaw > > Creo que eso contesta tu pregunta. > > Hola José. Gracias por tu contestación. Lo que me estas especificando el para hacer llamadas de salida (PEER). Yo necesito autentificar a un usuario de entrada, voy a intentar haciendo algo parecido solo cambiando a type=user para ver si así funciona. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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
[asterisk-users] Split asterisk in two ?? One TDM and One IP only??
I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am thinking of doing. Server1 Keep one physical server just for TDM functions. PRI to Telco PRI to old PBX for Fax. (basically using it as a mux) Keep meetme here for Digium card timing. Server2 Build a new asterisk install within Xen VM with data stored on an iSCSI SAN. This would be all IP. IAX and SIP extensions. IAX and SIP providers. IVR Voicemail Web access to voicemail CDR This way I can test different versions of the features of Server2 (clone with different IP) without affecting production. I assume that I just use an IAX or SIP trunk between the two asterisk servers. Does this make sense? Are others doing similar? Are there any other features that require the TDM card besides PRI, Fax and Meetme? I have heard of people using Xen for IP only asterisk, but are there any known gotchas? Thanks, Thank You, Steven BerkHolz ___ --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] [France CID] Does Zaptel support ETSI FSK?
On Mon, 22 Oct 2007 21:19:27 +0200, Vincent
<[EMAIL PROTECTED]> wrote:
>Does Zaptel support those on Digium TDM400 clones like those from
>OpenVox?
Pff, finally found what it was: It had nothing to do with zaptel, and
everything to do with extensions.conf:
exten => s,1,NoOp(Got a call)
;nothing displayed
exten => s,n,Verbose(${CALLERID})
exten => s,n,Verbose(${CALLERIDNAME})
exten => s,n,Verbose(${CALLERIDNUM})
exten => s,n,NoOp(${CALLERID})
exten => s,n,Verbose(${CALLERID})
;CID at last!
exten => s,n,Verbose(${CALLERID(num)})
I'm running Asterisk 1.4. Does someone know why only the last
statement does display the CID number while the others print nothing?
Thank you.
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Re: [asterisk-users] Extensions.conf for basic IVR?
On Mon, 22 Oct 2007 09:06:00 +0200, randulo <[EMAIL PROTECTED]> wrote: >The first ten sites that come up, including voip-info.org, usually a >good place to look first, each have full examples. Look also for the >background application wich is used to play the file, get input and >jump to the extension entered. Thanks. The problem with information on the Net is that the development of Asterisk moves quite fast, making some/a lot of information obsolete, something newbies aren't necessarily aware of. 2008 might be a good year to update "* - The future of telephony" :-) ___ --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] 16 ports wanted
Gergo Csibra schrieb: > Well, using more than one TDM card in your PC is not a good idea, > because of interrupts. If you have to have 16 FXO you can more > options: > > 1. Using TDM2400P with 4 FXO modules ($1775) > 2. Using Xorcom's Astribank (external) ($1170) > 3. Using some T1/E1 card with Channel Bank (more expensive) > 4. Using Sangoma's A200 with 8 (up to 12) dual-FXO modules (ca. $1.200) 5. Using Sangoma's A400 with 8 (up to 24) dual-FXO modules (ca. $1.350) Both Sangoma cards can be equipped with Octasic hardware-EC for ca. $300 more and are available in PCI(-X) and PCIexpress versions. For the A200 you need an additional case slot (does not need another PCI connector) for every 4 ports over 4. The same goes for the A400 on every 12 ports over 12. Christian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extensions.conf for basic IVR?
On 10/22/07, Vincent <[EMAIL PROTECTED]> wrote: > > 2008 might be a good year to update "* - The future of telephony" :-) Version 2 of TFOT was just released a few weeks ago... http://downloads.oreilly.com/books/9780596510480.pdf -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [France CID] Does Zaptel support ETSI FSK?
On Mon, 2007-10-22 at 23:18 +0200, Vincent wrote:
>
> exten => s,1,NoOp(Got a call)
>
> ;nothing displayed
> exten => s,n,Verbose(${CALLERID})
> exten => s,n,Verbose(${CALLERIDNAME})
> exten => s,n,Verbose(${CALLERIDNUM})
> exten => s,n,NoOp(${CALLERID})
> exten => s,n,Verbose(${CALLERID})
>
> ;CID at last!
> exten => s,n,Verbose(${CALLERID(num)})
>
>
> I'm running Asterisk 1.4. Does someone know why only the last
> statement does display the CID number while the others print nothing?
Beginning with Asterisk 1.4, we moved all of the CallerID functionality
from channel variables and applications to a single CALLERID dialplan
function. This should have been noted in UPGRADE.txt. I also tried to
warn you about it in my last email in this thread, but I guess I should
have been more specific.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Authenticate by IP?
On Mon, 2007-10-22 at 15:13 -0400, [EMAIL PROTECTED] wrote: > On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > > I have a customer that needs an Asterisk server to sell minutes for > > cell phones in Mexico. I do not see a problem with that since he will > > get the calls by SIP and then use GSM adapters to get the calls into the > > GSM network. My problem is that his customers only want to be > > identified by IP and not by a username and password. Is there a way to > > authenticate just by using an IP address? > > > > There certainly is. > And could you please point me in the right direction? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Extensions.conf for basic IVR?
On Mon, 22 Oct 2007 16:41:19 -0500, "Erik Anderson" <[EMAIL PROTECTED]> wrote: >Version 2 of TFOT was just released a few weeks ago... Just had to ask :-) Thanks. ___ --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] Authenticate by IP?
El Mon, Oct 22 de 2007 a las 15:59 -0500, Carlos Chavez comentaba: > Hola José. Gracias por tu contestación. Lo que me estas especificando > el para hacer llamadas de salida (PEER). Yo necesito autentificar a un > usuario de entrada, voy a intentar haciendo algo parecido solo cambiando > a type=user para ver si así funciona. type=peer also works for incoming calls. In this case (peer) asterisk only checks the IP the call is coming from and uses the context you defined there. If you use type=user you will need to specify a username and a secret. -- Greetings.. Víctor Toofic ___ --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] Authenticate by IP?
La configuración de Jose esta correcta. Cuando usas un "peer" en sip.conf Asterisk usa el hostname or el IP para autenticar. Cuando usas un "user" la autenticación se basa en el usuario y la contraseña, cual en su caso no existe. On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: > Hola José. Gracias por tu contestación. Lo que me estas > especificando > el para hacer llamadas de salida (PEER). Yo necesito autentificar a un > usuario de entrada, voy a intentar haciendo algo parecido solo cambiando > a type=user para ver si así funciona. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001 > > ___ > --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
[asterisk-users] bristuff: music on hold but no dialoptions tT defined.
Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Stopped music on hold on Zap/8-1 Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Started music on hold, class 'default', on channel 'Zap/8-1' Oct 22 11:20:55 VERBOSE[911] logger.c: == Spawn extension (macro-call, s, 2) exited non-zero on 'Zap /8-1' in macro 'tmp_call' ___ --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] [France CID] Does Zaptel support ETSI FSK?
On Mon, 22 Oct 2007 17:57:44 -0400, Jared Smith <[EMAIL PROTECTED]> wrote: >Beginning with Asterisk 1.4, we moved all of the CallerID functionality >from channel variables and applications to a single CALLERID dialplan >function. This should have been noted in UPGRADE.txt. I also tried to >warn you about it in my last email in this thread, but I guess I should >have been more specific. No problem. I should have read it more closely, but due to the number of people having problems with Zaptel and CID, I was focused on that part. Should have started asking people what the correct way was to read CID information in Asterisk 1.4... Thanks. ___ --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] G729a codecs + Asterisk 1.4.11
Dear Marc; I readed your email about the codec G729a and I am now also need to install the codec on my Asterisk. I typed from Asterisk CLI: core show version and I got the following: Asterisk SVN-branch-1.4-r72556 built by root @ localhost.localdomain on a i686 running Linux on 2007-06-30 13:08:08 UTC So I beleive that my processor is i686, correct? But I am not able to know which one to download: The x86-32 or x86-64 ? Can you please advise. Also, the nocona or the opteron versions? Regards Bilal --- Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI> show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] [France CID] Does Zaptel support ETSI FSK?
At 02:18 PM 10/22/2007, you wrote:
>;nothing displayed
>exten => s,n,Verbose(${CALLERID})
>exten => s,n,Verbose(${CALLERIDNAME})
>exten => s,n,Verbose(${CALLERIDNUM})
>exten => s,n,NoOp(${CALLERID})
>exten => s,n,Verbose(${CALLERID})
>
>;CID at last!
>exten => s,n,Verbose(${CALLERID(num)})
>
>
>I'm running Asterisk 1.4. Does someone know why only the last
>statement does display the CID number while the others print nothing?
try adding a wait(1) right in the beginning, worked for me.
Ira
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[asterisk-users] Voicemail playback on iPhone
Anyone managed to get this to work? What's the recipe? ___ --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
[asterisk-users] NAT traversal packet loss measurement
How can one measure the effect of NAT traversal packet loss? We currently have no solution for NAT traversal for our SIP clients. There is no doubt that packets are getting lost. What is not clear is how much damage this does. On the face of it, everything seems fine. Could this be so? Perhaps we're suffering a degradation in quality or our call setup times could be improved. How can we measure this? What's the simplest method of preventing packet loss due to NAT traversal in a SIP environment? Thanks, Yitzhak ___ --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] Voicemail playback on iPhone
Trick question I assume? It was mind numbingly simple on my iPhone...(though none of the voice mail worked when London a few weeks ago). - tap voice mail - - tap speaker (upper right) until it turns blue (is activate) - tap the message you want to playback - use assorted controls to delete - replay etc. Now...if the question is ... how do you get asterisk voice mail to show up on an iPhone...I am all ears. Groovy concept - if anybody has a hack - I'd love to see it. Elvis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Lixfeld Sent: Monday, October 22, 2007 4:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail playback on iPhone Anyone managed to get this to work? What's the recipe? ___ --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 __ NOD32 2607 (20071022) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --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] NAT traversal packet loss measurement
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Yitzhak Bar Geva wrote: > How can one measure the effect of NAT traversal packet loss? > We currently have no solution for NAT traversal for our SIP clients. There > is no doubt that packets are getting lost. What is not clear is how much > damage this does. On the face of it, everything seems fine. Could this be > so? Perhaps we're suffering a degradation in quality or our call setup times > could be improved. How can we measure this? > What's the simplest method of preventing packet loss due to NAT traversal in > a SIP environment? NAT is unlikely to cause a percentage of packets to get lost. Normally you'd have one way audio if NAT was causing a problem (i.e. 100% packet loss). The only other situation in which it might happen is where the NAT router decides to close a port mapping (thereby blocking incoming calls to the customer's device). But if you're looking for packet loss there are a number of other things to check first. I wouldn't do VoIP across the WAN without at least some packet shaping but hey. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHUC+DQNt8rg0Kp4RAv0uAJ9Q41eQ+7RuqzFvgtxEhQOIU0QFggCaAlkD GMVdY/n58wHsciuHihZCCHY= =6L87 -END PGP SIGNATURE- ___ --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
[asterisk-users] Force codec order
There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. ___ --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] Snom 360 lights not working on subscription
What we found is that even if you get the lights working, they go off after a few days. Paul Hales AsteriskIT On Mon, 2007-10-22 at 09:49 -0300, Carlos Maimone wrote: > Dear friends, > > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX > > In order to get subscriptions working and the Snom 360 lights turns > on, I have set everything just like all the pages in the net explain. > > So, I get subsciption working. I can list subscription on the > asterisk and if I use the SIP trace function built in at the SNOM nad > see NOTIFY messages and 200 OK responses. But I realized that content > length = 0 in all messsages and there isn't any XML content in those > Notify headers.. > > > any idea of what's going on? > > IN SNOM 360 I am currently using firmware 6.5.12 > > I am pretty sick dealing with this issue. > > > thanks and regards, > > > Charlie > > ___ > --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] Authenticate by IP?
Carlos, No solo para enviar llamadas, sino tambien para recibir (de hecho, ese bloque que puse ahi lo uso para recibir, no para enviar). Te posteo un ejemplo del ejemplo que trae asterisk de sip.conf ;[sip_proxy] ; For incoming calls only. Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) ;type=peer ;context=from-fwd ;host=fwd.pulver.com Como ves, para validar por IP se usa en el campo 'host' el ip o dominio de quien nos enviara la llamada :) -- Jose P. Espinal slackware-es.com Carlos Chavez wrote: On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote: Saludos Carlos, Como vas a recibir las llamadas via SIP, puedes especificar el IP del host que te enviara las llamadas, por ej. Este es un bloque que tengo definido en el SIP.conf de uno de mis servers para enrutar las llamadas internacionales y a telefonos moviles utilizando un proveedor de terminacion. [oficina] type=peer context=from_office ; Esto va a mi 'extensions.conf' host=200.88.42.29; Este es el ip publico en la oficina (estatico) nat=no canreinvite=no qualify=yes disallow=all allow=g729 allow=ulaw Creo que eso contesta tu pregunta. Hola José. Gracias por tu contestación. Lo que me estas especificando el para hacer llamadas de salida (PEER). Yo necesito autentificar a un usuario de entrada, voy a intentar haciendo algo parecido solo cambiando a type=user para ver si así funciona. ___ --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] Snom 360 lights not working on subscription
I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote: > Dear friends, > > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX > > In order to get subscriptions working and the Snom 360 lights turns > on, I have set everything just like all the pages in the net explain. > > So, I get subsciption working. I can list subscription on the > asterisk and if I use the SIP trace function built in at the SNOM nad > see NOTIFY messages and 200 OK responses. But I realized that content > length = 0 in all messsages and there isn't any XML content in those > Notify headers.. > > > any idea of what's going on? > > IN SNOM 360 I am currently using firmware 6.5.12 > > I am pretty sick dealing with this issue. > > > thanks and regards, > > > Charlie > > ___ > --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] Video Conference
CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at http://sourceforge.net/projects/webhuddle It has built in VOIP of some kind, don't remember the details. But why not use Asterisk or one of the free teleconference websites for the audio and WebHuddle for the webcams and desktop sharing. On 10/22/07, John Millican <[EMAIL PROTECTED]> wrote: > > Hello All, > I am looking at doing some video conferencing with SIP. I was hoping to > get > some early pointers from any one that is currently doing this. I have > been > all over goggle and voip-info and there is a ton of anecdotal information > but, I was hoping for more specifics of what people are actually using > that > works and even some of what hasn't worked so that I can stay away. What I > am > considering at this point is hacking up my own solution using off the > shelf > equipment. Decent web camera, Polycom conference phone(maybe if the > budget > holds) and a large wide screen LCD monitor all connected to * > Sound reasonable or am I living a pipe dream? > JohnM > > > ___ > --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] Snom 360 lights not working on subscription
I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote: > Dear friends, > > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX > > In order to get subscriptions working and the Snom 360 lights turns > on, I have set everything just like all the pages in the net explain. > > So, I get subsciption working. I can list subscription on the asterisk > and if I use the SIP trace function built in at the SNOM nad see > NOTIFY messages and 200 OK responses. But I realized that content > length = 0 in all messsages and there isn't any XML content in those > Notify headers.. > > > any idea of what's going on? > > IN SNOM 360 I am currently using firmware 6.5.12 > > I am pretty sick dealing with this issue. > > > thanks and regards, > > > Charlie > > ___ > --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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --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] 16 ports wanted
What do you mean by interruption? Is it possible to better control to prevent it? The options you provided is over my budget. That's why I am looking for multiple TDM cards. On 10/22/07, Gergo Csibra <[EMAIL PROTECTED]> wrote: > Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote: > > > Hi all, > > I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 > > or 2 TDM808 to get 16 FXO? What is the difference (in performance and > > control) in using 4 x TDM404 and 2 x TDM808 if possible? > > ango > > Well, using more than one TDM card in your PC is not a good idea, > because of interrupts. If you have to have 16 FXO you can more > options: > > 1. Using TDM2400P with 4 FXO modules ($1775) > 2. Using Xorcom's Astribank (external) ($1170) > 3. Using some T1/E1 card with Channel Bank (more expensive) > > > -- > Best regards, > Gergomailto:[EMAIL PROTECTED] > > > ___ > --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] Snom 360 lights not working on subscription
Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: > I also have problems with these phones. I have deployed many of them > and have had nothing but problems. Omar, what phones did you switch to? > I needed some of the features of the snom phones, like the multiple > buttons with prescence lights. > > Mike > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. > Sabek > Sent: Monday, October 22, 2007 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Snom 360 lights not working on > subscription > > I used to deploy these phones, it was these types of issues that forced > me to drop it. It took way too long to troubleshoot the problems and > there was a general lack of documentation. This was 2 years ago, things > might have changed. If I remember correctly, it was this issue you are > having that was the final straw. > > Good luck, > > Omar > > On 10/22/07, Carlos Maimone <[EMAIL PROTECTED]> wrote: > > Dear friends, > > > > I am working around with a Snom 360 and Asterisk 1.4 + FreePBX > > > > In order to get subscriptions working and the Snom 360 lights turns > > on, I have set everything just like all the pages in the net explain. > > > > So, I get subsciption working. I can list subscription on the asterisk > > > and if I use the SIP trace function built in at the SNOM nad see > > NOTIFY messages and 200 OK responses. But I realized that content > > length = 0 in all messsages and there isn't any XML content in those > > Notify headers.. > > > > > > any idea of what's going on? > > > > IN SNOM 360 I am currently using firmware 6.5.12 > > > > I am pretty sick dealing with this issue. > > > > > > thanks and regards, > > > > > > Charlie > > > > ___ > > --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 > > > This E-mail, including any attachments, may be intended solely for > the personal and confidential use of the sender and recipient(s) named > above. This message may include advisory, consultative and/or > deliberative material and, as such, would be privileged and confidential > and not a public document. Pursuant to 42 CFR, any information in this > e-mail identifying a former, present, or potential client of Straight & > Narrow is confidential. If you have received this e-mail in error, you must > not review, transmit, convert to hard copy, copy, use or disseminate this > e-mail or any attachments to it and you must delete this message. You are > requested to notify the sender by return e-mail. > > > ___ > --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] 16 ports wanted
Rilawich Ango wrote: > What do you mean by interruption? Is it possible to better control to > prevent it? The options you provided is over my budget. That's why > I am looking for multiple TDM cards. > > On 10/22/07, Gergo Csibra <[EMAIL PROTECTED]> wrote: >> Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote: >> >>> Hi all, >>> I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 >>> or 2 TDM808 to get 16 FXO? What is the difference (in performance and >>> control) in using 4 x TDM404 and 2 x TDM808 if possible? >>> ango >> Well, using more than one TDM card in your PC is not a good idea, >> because of interrupts. That's what he meant: http://en.wikipedia.org/wiki/Interrupt http://en.wikipedia.org/wiki/Interrupt_request http://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html btw: Quoting signatures is a bad habit. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] 16 ports wanted
The Xorcom Astribanks are quote good - have you looked at those? PaulH On Tue, 2007-10-23 at 12:41 +0800, Rilawich Ango wrote: > What do you mean by interruption? Is it possible to better control to > prevent it? The options you provided is over my budget. That's why > I am looking for multiple TDM cards. > > On 10/22/07, Gergo Csibra <[EMAIL PROTECTED]> wrote: > > Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote: > > > > > Hi all, > > > I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404 > > > or 2 TDM808 to get 16 FXO? What is the difference (in performance and > > > control) in using 4 x TDM404 and 2 x TDM808 if possible? > > > ango > > > > Well, using more than one TDM card in your PC is not a good idea, > > because of interrupts. If you have to have 16 FXO you can more > > options: > > > > 1. Using TDM2400P with 4 FXO modules ($1775) > > 2. Using Xorcom's Astribank (external) ($1170) > > 3. Using some T1/E1 card with Channel Bank (more expensive) > > > > > > -- > > Best regards, > > Gergomailto:[EMAIL PROTECTED] > > > > > > ___ > > --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 ___ --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] dial-out call queue
If you want to do this automatically, what you're looking for is a (Predictive) Dialler for Asterisk. There are a few available, both on the commercial and the free side. I'd start by checking out ViciDial /free) and SineDialer (commercial) that are some of the most used ones. Thanks l. On Mon, 22 Oct 2007 22:57:47 +0200, Joao Pereira <[EMAIL PROTECTED]> wrote: > Is it possible to implement a dial-out call queue in Asterisk? > My idea is to give Asterisk a list of numbers, and then he makes the > calls and delivers the calls to a call queue. > Then, the agents will answer the calls. > Is this possible? > Thanks > Regards > Joao pereira > -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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
[asterisk-users] CFP for HITBSecConf2008 - Dubai now open
The CFP for HITBSecConf2008 - Dubai is now open. Our 2008 event is expected to attract over 300 attendees from around the EMEA region and will see keynote speakers Bruce Schneier (Founder and CTO, BT Counterpane) and Jeremiah Grossman (Founder and CTO, White Hat Security). The event is supported and endorsed by the UAE Telecommunications and Regulatory Authority. Being a deep-knowledge technical conference, talks that are more technical or that discuss new and never before seen attack methods are of more interest than a subject that has been covered several times before. Summaries not exceeding 250 words should be submitted (in plain text format) to [EMAIL PROTECTED] for review and possible inclusion in the programme. Submissions are due no later than 1st January 2008. Topics of interest include, but are not limited to the following: # 3G/3.5G/4G Cellular Networks # Apple / OS X vulnerabilities # SS7/Backbone telephony networks # Smart Card Security and Biometric Systems # UMTS, HSDPA, GPRS and CDMA Security # Security of Wimax, WLAN, Bluetooth, GPS and other wireless technology # Analysis of network and security vulnerabilities # Firewall and Intrusion detection technology # Data Recovery and Incident Response # Network Protocol and Analysis # Analysis of malicious code # Applications of cryptographic techniques # Analysis of attacks against networks and machines # File system security PLEASE NOTE: We do not accept product or vendor related pitches. If your talk involves an advertisement for a new product or service your company is offering, please do not submit. Your submission should include: # Name, title, address, email and phone/contact number # Draft of the proposed presentation (in PDF or PowerPoint format), proof of concept for tools and exploits, etc. # Short biography, qualification, occupation, achievement and affiliations (limit 150 words). # Summary or abstract for your presentation (limit 250 words) # Time (max 60 minutes including time for discussion and questions) # Technical requirements (video, internet, wireless, audio, etc.) Each non-resident speaker will receive accommodation for 2 nights/ 3 days. For each non-resident speaker, HITB will cover travel expenses up to USD 1,000.00. HITBSecConf2008 - Dubai http://conference.hitb.org/hitbsecconf2008dubai/ HITBSecConf 2003,2004,2005,2006 Videos are now on Google Video http://www.hackinthebox.org/modules.php?op=modload&name=News&file=article&sid=24719&mode=thread&order=0&thold=0 ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Jared Smith wrote: > On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote: >> The SPA-9x1 does support http download, but I don't see how you could >> change the initial TFTP request to HTTP without manually configuring >> the phone. Even then I'm not sure it would work - I certainly >> haven't managed to make any of my SPAs do an auto-config over HTTP. > > Actually, it's really easy to do. Here's a copy of my spa942.cfg file > which I use to point the phone at my web server, as well as upgrade > the firmware. Now that I've found my typo, I completely agree :-) I had $MAC instead of $MA, which produces a MAC address in the nn:nn:nn format. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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] NAT traversal packet loss measurement
Yitzhak Bar Geva wrote: > How can one measure the effect of NAT traversal packet loss? > We currently have no solution for NAT traversal for our SIP clients. We've recently completed a setup (see other threads) with a couple of SIP clients behind NAT in their respective home-offices. Took a couple of attempts, but after consulting the list, we have a working setup. > What's the simplest method of preventing packet loss due to NAT > traversal in a SIP environment? I doubt very much if any loss you're seeing is due to NAT traversal. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --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
