RE: [asterisk-users] Add Apps to Asterisk?
First, check for app_meetme.so in /usr/lib/asterisk/modules (wherever your modules path is). Next, in the CLI, do a 'show modules' to see if it is there. If not, check your modules.conf and add in 'load => app_meetme.so' assuming autoload is not enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Tuesday, November 14, 2006 21:09 To: Asterisk-Users Subject: [asterisk-users] Add Apps to Asterisk? I've got an Asterisk (v1.2.11) installation running, but it doesn't seem to have the Meetme() app. At the CLI, I type Meetme , and it responds No such command 'Meetme'; meetme doesn't show up in CLI show modules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There is an /etc/asterisk/meetme.conf file, but I don't see anything to use it. What do I have to do, exactly, to install Meetme? What about the Conference command, or others not installed? I'd prefer to use the CentOS package system as much as possible, but I can compile source if necessary. Is there a HowTo on the Web somewhere that details this process? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS
Hey!!! I spent an ENTIRE Day googling for this issue only to find your single solitary response (& solution) here on the mailing list. How in the heck did you know to do that? and as popular as IIS (is) I wonder why the answer was so hard to find on the web. If you have time, please let me know! how in the world you got that answer. It was very frusterating that there's so little information about this problem or it's cause... I spent all day trying things & looking for answers... added the mime type & all for .xhtml to no avail... Thanks! Steve > I found this solution from the web and figured I'd share it because it > affects all phones getting input from IIS. > > Map .gif, .jpg, .css etc (in my case I used .xhtml for the Polycom 601) > in IIS under your sites: > > Properties -> Virtual directory tab-> Configuration -> Application > configuration -> Mappings tab. > Make ASP DLL [..\inetsrv\asp.dll] to handle these files. > > This allows the file with extension XHTML to be passed to the phone and > not return a HTTP 406 error (File type not supported by your browser). > > > Hope is helps others. > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Phil > Menico > Sent: Friday, August 25, 2006 8:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 > withIIS > > > Thanks, but we have reasons to want to make it work with IIS. > > Anyone have a hint of what is the issue? > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Douglas > Garstang > Sent: Thursday, August 24, 2006 6:46 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Polycom microbrowser issue Error HTTP 406 > withIIS > > > > We had a similar problem. Eventually we gave up and just used apache. We > found that _exactly_ the same content would not work with IIS, but WOULD > work with Apache. > > -Original Message- > From: Phil Menico [mailto:[EMAIL PROTECTED] > Sent: Thursday, August 24, 2006 3:06 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Polycom microbrowser issue Error HTTP 406 with > IIS > > > I have no where else to turn to so if anyone has an answer please send > it. > > I am running sip version 1.6.on a Polycom 601 on Asterisk and am unable > to get the microbroser to work. The phone returns a 406 error for both > idle and services. > I can see the file being requested and the subsequent 406 error in the > IIS log files. > Any ideas on what permissions are needed in IIS or how to format the > webpage file? > I tried both these 2 files with no luck > > XHTML file 1: > > > > > > Hello phil post > > > > > XHTML file 2: > > > http://www.w3.org/1999/xhtml"; xml:lang="en" lang="en"> > > Virtual Library > > > Hello phil > > > > Log info from IIS: > > 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET > /Polycom/ - 302 0 295 202 0 HTTP/1.1 10.0.1.210:81 > Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPoin > tIP-SPIP_601)+libcurl/7.12.1 - - > 2006-08-24 20:39:18 10.0.3.175 - W3SVC1 PHIL3 10.0.1.210 81 GET > /Polycom/post.htm - 406 0 4085 242 10 HTTP/1.1 10.0.1.210:81 > Polycom-Microbrowser/1.0+(SIP/1.6.3.0067;+SoundPoint+IP+PolycomSoundPoin > tIP-SPIP_601)+libcurl/7.12.1 - http://10.0.1.210:81/Polycom > > > > > Thank you. > > Phil > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retain call control: Avoid letting call get
Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go (like asterisk vm, or ...) So it does what you are asking, if you don't care for freepbx, you can look at the dialplan it generates and get some good ideas. (It basically does what Dovid mentions). philippel From: "Dovid B" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Date: Wed, 15 Nov 2006 02:36:53 +0200Subject: Re: [asterisk-users] Retain call control: Avoid letting call getintocellular voicemail You can create a macro that the person that is called has to press a key to take the call. If no key is pressed then you can send them to a menu where they can press one to leave a message or two to go back to the menu.On 11/14/06, joe a. <[EMAIL PROTECTED]> wrote: Did not know how to make up a subject line for this.I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible?joe a.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sponsored LinkMortgage rates near 39yr lows. $420,000 Mortgage for $1,399/mo - Calculate new house payment___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1&SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten => s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten => s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten => s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten => s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten => s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten => s,1,SetGlobalVar(RingTimeout=35)
exten => s,2,NoOp("${CALLERID}")
exten => s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},"")
On Tue, 14 Nov 2006, Larry Alkoff wrote:
> My SIP phones can dial out through Sipura SPA3k to POTS for local and
> 911 calls _but_ incoming POTS calls are being swallowup somehow.
>
> Am I on the right track with the code snippit below?
>
> sip.conf:
> -
> In sip.conf the following code is _supposed_ to ring the SIP phones when
> a POTS line call comes in through Sipuara to Asterisk.
>
> [spa3k-pstn-in] ; Pots-line-in from Sipura
> ; If you're using Asterisk, this goes into the Incoming settings
> ; For your Trunk
> host=dynamic
>
> type=friend ; should be peer if incoming only ??
>
> context=[macro-ringall] ;ring all the sip phones
>
> secret=x
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> insecure=very
>
>
> extensions.conf
>
> context to ring all SIP phones when a POTS call comes into SPA3k:
>
> [macro-ringall] ; ring all SIP phones
> exten => s,1,Dial(SIP/120&SIP/121&SIP/122&SIP/124&SIP/125&SIP/126&SIP/127)
> exten => s,2,hangup
>
> --
> Larry Alkoff N2LA - Austin TX
> Using Thunderbird on Linux
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Re: [asterisk-users] trixbox + agi
If I were you I would go the AGI way. Use ruby, python, php, perl, java, c# or even erlang. Aything but the asterisk dialplan commands. There is no sense in putting yourself through that pain. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDD - stops receiving characters
We are working on a TDD application and testing with an Ultratec SuperPrint 4425 in Australia. We've got it to exchange characters fairly cleanly for a minute or two but after that we seem to be getting a signal but no character in Asterisk. Has anybody done any work with the Asterisk TDD functionality and got it to work reliably? Does anyone have any idea what might be causing the system to stop receiving characters after a particular period of time? Warrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller Initiated Conference
Hi ALL, configuring the caller initiated conference, any one has idea about this please replay me back... -Linga Reddy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Add Apps to Asterisk?
I've got an Asterisk (v1.2.11) installation running, but it doesn't seem to have the Meetme() app. At the CLI, I type Meetme , and it responds No such command 'Meetme'; meetme doesn't show up in CLI show modules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There is an /etc/asterisk/meetme.conf file, but I don't see anything to use it. What do I have to do, exactly, to install Meetme? What about the Conference command, or others not installed? I'd prefer to use the CentOS package system as much as possible, but I can compile source if necessary. Is there a HowTo on the Web somewhere that details this process? -- (C) Matthew Rubenstein ___ --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] trixbox + agi
On 11/14/06, Tim Uckun <[EMAIL PROTECTED]> wrote:
On 11/15/06, blackwater dev <[EMAIL PROTECTED]> wrote:> I need to write an app which takes a phone call, asks for the user to input> a number and then queries a db via a webservice and reads the results a row
> at a time back to the caller. First, is this beyond Asterisk? Second, can> I do this if I use the Trixbox implementation? Third, any good tutorials on> doing just this?There are numerous AGI toolkits in different languages. I have just
started fooling around with RAGI which is integrated with ruby onrails. From my experiments so far it seems to work OK.___--Bandwidth and Colocation provided by
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I do this currently but dont know how to allow additional rows to be
read back. It only reads the first one then stops. Any insight on how
to have it loop to get addtional rows would be great. Also I was wondering how to get DTMF Digits to translate in to possible matches for LastName to query my database like the Directory CMD does.
[custom-lookup]
exten => s,1,Cepstral(Please enter your eye dee Number, followed by the pound key now.)
exten => s,2,Read(ID)
exten => s,3,MYSQL(Connect connid localhost contacts contacts)
exten => s,4,MYSQL(Query resultid ${connid} SELECT\
FirstName\,LastName\,HomePhone\ FROM\ contacts\ WHERE\
ContactID=\'${ID}\')
exten => s,5,MYSQL(Fetch foundRow ${resultid} var1 var2 var3) ; fetch row
exten => s,6,GotoIf($["${foundRow}" = "1"]?7:9) ;
exten => s,7,Cepstral(The Phone number for ${var1} ${var2} is, ${var3}.)
exten => s,8,Goto(s,10)
exten => s,9,Cepstral(No match found. Goodbye) ; End loop if
exten => s,10,MYSQL(Clear ${resultid})
exten => s,11,MYSQL(Disconnect ${connid})
If you need more help with the above let me know.Tom Vile
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Re: [asterisk-users] trixbox + agi
On 11/15/06, blackwater dev <[EMAIL PROTECTED]> wrote: I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row at a time back to the caller. First, is this beyond Asterisk? Second, can I do this if I use the Trixbox implementation? Third, any good tutorials on doing just this? There are numerous AGI toolkits in different languages. I have just started fooling around with RAGI which is integrated with ruby on rails. From my experiments so far it seems to work OK. ___ --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: [SPAM HEADER] - [asterisk-users] trixbox + agi - Email found in subject
Lumenvox recently put out a press release regarding integration of their technology with Trixbox http://www.lumenvox.com/news/lumenvoxNews/2006/092506.aspx If they have a TTS engine as part of this integration, you should be able to create an IVR with Trixbox to grab DTMF input from a user, query a local or remotely accessible database, and run the query result through a TTS engine and output to the caller. In terms of a traditional IVR application, it's a fairly simplistic app, but I am not sure how far along Lumenvox is with Trixbox integration. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of blackwater devSent: Tuesday, November 14, 2006 10:15 PMTo: asterisk-users@lists.digium.comSubject: [SPAM HEADER] - [asterisk-users] trixbox + agi - Email found in subject I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row at a time back to the caller. First, is this beyond Asterisk? Second, can I do this if I use the Trixbox implementation? Third, any good tutorials on doing just this? Thanks! ___ --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] Desktop integration
Does your class handle the reputed failure of the Manager interface to originate some calls under heavy load? Does anyone know if that unreliable feature is fixed in newer Asterisk versions, or in 1.4? On Tue, 2006-11-14 at 18:47 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 15 Nov 2006 03:24:20 +0200 > From: "Dovid B" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Desktop integration > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > reply-type=response > > > > > I've written a java program (using asterisk-java) that invokes the > > originate method on the manager API. > > which can be put in an CGI. If anyone is interested I can clean it > up and > > post it somewhere. > > > I am sure ther are a lot of interested users. Why not put it up on the > wiki > ? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trixbox + agi
I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row at a time back to the caller. First, is this beyond Asterisk? Second, can I do this if I use the Trixbox implementation? Third, any good tutorials on doing just this? Thanks! ___ --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] Call log reveals redundant calls!
- Original Message - From: "Mike Benjamin" <[EMAIL PROTECTED]> To: Sent: Wednesday, November 15, 2006 3:47 AM Subject: [asterisk-users] Call log reveals redundant calls! Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773 "8183461773" <8183461773> 1 NO ANSWER 2 48. 2006-11-14 16:40:55 Local/8183... 8183461773 "Valley Heme Onc" <8183461773> 8183461773 NO ANSWER 4 49. 2006-11-14 16:40:54 IAX2/Voice... 8183461773 "8183461773" <8183461773> 1 NO ANSWER 5 Thanks, Dr. B. For the future when starting a post please dont repost a previous message. Do you have your outbound dialing extension in the context that is set to default in sip.conf ? ___ --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] Installation of Unicall for MFC/R2
Hi Moises, > 1. On what directory/folder should I copy the chan_unicall.c,> channels_makefile.patch?into wherever you had put the source code of your asteriskinstallation. There, you must have a folder named "channels", whereyou will see several files named chan_sip, chan_zap and in generalchan_xxx, there you should put chan_unicall.c Thank you very for this information, this will be a good start for me to search for this. > 2. On what directory/folder should I command patch <> channels_makefile.patch?You need to learn somewhere else how to apply patches. Yeahhh, that's why I'am searching the WorlWideWeb. > 3. On what directory/folder should I command make and make install?God, I think you need a basic linux install course, read in googleabout "Makefiles" Frankly speaking, YES, I need training for Linux Install. But before going to formal training I need to make this * box up & running ASAP before my CIO fired me. I just thought learning through forums is faster and easier because we are guided based on actual experience. In fact, this is the reason I ended up in this forum because of my persistence to learn in a fastest way and of course in a most economical way, reason why we are into Open Source. (Savings). > 4. What actually the asterisk do with patch < channels_makefile.patch?Apply the patch to the Makefile code I am really very thankful to all of you guys, hope I would be able to help in some other subject matter and rest assured, I will post of any successful development with my project. > Desperately need help.Everyone does. Thanks God, there are FORUMS as our references. Cheers! Angel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call log reveals redundant calls!
Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773 "8183461773" <8183461773> 1 NO ANSWER 2 48. 2006-11-14 16:40:55 Local/8183... 8183461773 "Valley Heme Onc" <8183461773>8183461773 NO ANSWER 4 49. 2006-11-14 16:40:54 IAX2/Voice... 8183461773 "8183461773" <8183461773> 1 NO ANSWER 5 Thanks, Dr. B. --- [EMAIL PROTECTED] wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > >1. Re: Load balance Asterisk servers? (David Thomas) >2. Re: config template for Grandstreams (Bob Chiodini) >3. RE: 900 rules (Ron McLeod) >4. Re: Retain call control: Avoid letting call get into cellular > voicemail (was: Dialplan options) (Anselm Martin Hoffmeister) >5. Re: Dialing from "Placed Calls" on Polycom IP501doesn't > always work (Anthony Rodgers) >6. Re: Polycom - how to 'buddy watch' trunks? (Anthony Rodgers) >7. DUNDi Asterisk Cluster (David Thomas) >8. Re: OutCall Release (Tzafrir Cohen) >9. Re: Survey: In what ways do you use Asteriskat your house? (Ira) > 10. In the beginning-The first question. (James R. Stevens) > 11. RE: Problem found Re: [asterisk-users] Headaches with Video > over SIP (Peter Howard) > 12. Re: In the beginning-The first question. (Doug Lytle) > 13. RE: OutCall Release (Senad Jordanovic) > 14. Re: Problem with FXS ports of TDM400P (Gustavo Felisberto) > 15. unable to get channel lock BAD BAD BAD (Tim Uckun) > 16. Re: Load balance Asterisk servers? (Aaron Daniel) > 17. Re: "Username/auth name mismatch" + SIP phone can't connect? > (Fred) > 18. Re: DUNDi Asterisk Cluster (Aaron Daniel) > 19. How do I change the rtp packet size in a Cisco 7490 from 10ms > to 20ms (Naija Man) > 20. Re: Problem with FXS ports of TDM400P (Gustavo Felisberto) > 21. Voice mail transfer between 2 asterisk servers (Naija Man) > 22. RE: Problem with FXS ports of TDM400P (Robert Jenkins) > 23. RE: In the beginning-The first question. (James R. Stevens) > 24. Re: In the beginning-The first question. (Doug Lytle) > 25. Re: Re: Is asterisk able to integrate with MS SQL (Sharon Lim) > > > -- > > Message: 1 > Date: Tue, 14 Nov 2006 12:00:25 -0700 > From: "David Thomas" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Load balance Asterisk servers? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 11/14/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: > > Incorrect :) IAX2 most definitely does support regcontext. > > > > Also, I think what he means is the phone specific information must be > > exactly the same from system to system or the failover won't be as > > seamless as you expect. A lot of phones support some sort of SRV > > records, so in the event of a failure, the phones will automatically > > find the next available server. The other option there is to set up an > > HA environment so the failover is even transparent to the phones, they > > just start talking to the new IP address immediately. > > > > Another thought, in any failover situation, if you have any sort of > > automated failover, you must make sure phones that need specific > > features fail to the same server (i.e. hinting and such) as those > > features don't work cross server. > > > > Aaron > > > > On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: > > > On 11/14/06, Stelios Koroneos <[EMAIL PROTECTED]> wrote: > > > > JR Richardson gave a very nice presentation at Astricon on how to do > that with DUNDI > > > > > > As I understand it JR Richardson's DUNDi solution does not support > > > IAX. It uses regcontex which I believe is only available with SIP. > > > (please correct me if I'm wrong) > > > > > > Also JR notes that... > > > > > > "Associated SIP Users, business customers, require same registration > > > and failover to the same servers" so unless this is for a residential > > > setup, it may not be of much use to you. Nevertheless it
Re: [asterisk-users] Installation of Unicall for MFC/R2
1. On what directory/folder should I copy the chan_unicall.c, channels_makefile.patch? into wherever you had put the source code of your asterisk installation. There, you must have a folder named "channels", where you will see several files named chan_sip, chan_zap and in general chan_xxx, there you should put chan_unicall.c 2. On what directory/folder should I command patch < channels_makefile.patch? You need to learn somewhere else how to apply patches. 3. On what directory/folder should I command make and make install? God, I think you need a basic linux install course, read in google about "Makefiles" 4. What actually the asterisk do with patch < channels_makefile.patch? Apply the patch to the Makefile code Desperately need help. Everyone does. Best Regards and good look Moises Silva -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ --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] Problem with FXS ports of TDM400P
Dovid B wrote: > Was the phone off hook when the card was started ? Also as some one > mentioned above is there power being provided to the card ? > Thre was no phone connected to the card when the module was loaded. Is that needed? The card has the power connector plugged in. -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reception Console
The reception console I mentioned a few weeks ago is out and about. Much more information is available on the wiki: http://www.voip-info.org/wiki/index.php?page=The-Receptionist-Console later, PaulH ___ --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] Desktop integration
I've written a java program (using asterisk-java) that invokes the originate method on the manager API. which can be put in an CGI. If anyone is interested I can clean it up and post it somewhere. I am sure ther are a lot of interested users. Why not put it up on the wiki ? ___ --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] Can I disable send e-mail feature in the voicemailapplication?
Yes. Just dont add a line in voicemail.conf with an email address. - Original Message - From: "Ma Zhiyong" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, November 14, 2006 1:34 PM Subject: [asterisk-users] Can I disable send e-mail feature in the voicemailapplication? HI, all Can I disable send e-mail feature in the voicemail application? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Wan Router with Failover
I have not seen anything yet that does failover based on SIP (i.e. seeing if there is a lag with sip). Anyone know if that exists ? - Original Message - From: Dovid B To: asterisk-users@lists.digium.com Sent: Tuesday, November 14, 2006 3:49 AM Subject: [asterisk-users] Dual Wan Router with Failover Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA with reliable FAX?
Curt Shaffer wrote: I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I’m looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. (a) If you are not running a version of Asterisk that has working SIP jitter buffering (is there such a thing?), then abandon all hope now. (b) We have no experience with the Cisco ATAs, but the Linksys (nee Sipura) SPA-210x is markedly better than the SPA-100x and SPA-200x, probably because they have better jitter buffering (it goes without saying we do not pass our fax traffic through Asterisk). (c) T.38 is the way to go, G.711 a poor and distant second choice (again, Asterisk's T.38 pass-through is far from ready for prime time). g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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] Re: Grandstream TFTP system wide settings
Perhaps you want to look at http://tanesha.net/Wiki/GratissipTftpd.html You can keep the P-codes in a mysql database and build all the configs you want. For me, it was a little too much work for the few phones I have, but if you need more. Todd Is there a quicker way to change settings for all Grandstream phones, is there any one file which can act as a global configuration file without changing each phones phone specific settings? And can't it be simply done by text editing, without the need to convert each file to cfg format? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA with reliable FAX?
I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I’m looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using in this type of setup with the most luck? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DUNDi Asterisk Cluster
> I would set up a separate set of boxes that can directly connect to the providers. That gives you separation of services, and if one of those boxes goes down, you'll have enough saturation to not have to worry about it. Having the main boxes do the connections to the providers adds a level of complexity when you need to bring a new system online. Just my 2 cents :) -- Aaron Daniel Use 2 failover dundi servers within the cluster, point those servers to the service provider, all other asterisk servers reinvite to the service provider through the dundi servers. The service provider has to allow re-invites, but only authorized from the dundi servers. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: - In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If you're using Asterisk, this goes into the Incoming settings ; For your Trunk host=dynamic type=friend ; should be peer if incoming only ?? context=[macro-ringall] ;ring all the sip phones secret=x dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very extensions.conf context to ring all SIP phones when a POTS call comes into SPA3k: [macro-ringall] ; ring all SIP phones exten => s,1,Dial(SIP/120&SIP/121&SIP/122&SIP/124&SIP/125&SIP/126&SIP/127) exten => s,2,hangup -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] Can AGI do this?
Bret Schuhmacher, who happens to be smarter than you, thinks:
> Please pardon the absolute noob questions. Someone has asked me to
> interface with Asterisk and have it dial 4 numbers in succession
I dont understand this:
> to have
> it track down an on-call person.
> If not, is it possible to write an AGI program that gets all 4 numbers,
> then somehow hands them one-by-one to Asterisk? If so, how does
> Asterisk manage the communication of "failed to complete the call" with
> the AGI app? Does the AGI just monitor stdin looking for status
> messages and returns the next number?
>
> If Asterisk/AGI can do both, is the first method better than the
> second? It certainly seems easier.
The AGI script can die when it finished they job, probably I'm
missunderstanding what you want but maybe this examples helps:
exten => 123,1,Answer
exten => 123,n,AGI(somescript)
exten => 123,n,Dial(${TRUNK}/${FOURDIGITNUMBER})
exten => 123,n,Hangup
somescript: just sends AGI command SETVAR FOURDIGITNUMBER 2468
[context]
exten => s,1,Answer
exten => s,n... (set timeouts, et all).
exten => s,n,Background(audio)
exten => s,n,WaitExten
exten => _,1,Dial(${EXTEN})
exten => t,..
exten => h,...
--
Onward through the fog.
___
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Re: [asterisk-users] Defunct / zombie AGI after some execution time
Does asterisk close the channel?On 11/13/06, Mark <[EMAIL PROTECTED]> wrote: Hello,We are running Asterisk-1.0.12 in a CentOS 4-4.2 system, kernel2.6.9-42.0.3.ELsmp.We have some custom AGI, and when we launch Asterisk the system works fine.But **after some time**, each AGI execution generates a zombie process. We believe that it's not a problem in the AGI code, because Asterisk+AGI isworking fine in the first "n" minutes/hours. This is a pstree sample:init-+-asterisk---asterisk---48*[asterisk] But after some execution time, this is the pstree output:init-+-asterisk---asterisk-+-28*[asterisk] | |-asterisk-+-21*[x.agi] | | `-40*[x.agi ] | |-5*[asterisk-+-y.agi] | | |-z.agi](...)Each agi is a defunct process. It dies when the call (parent) finishes.When the first zombie appears, then ALL next AGI launched from Asterisk generates a zombie.We have tested some improvements to solve the problem, with no success:- Upgrade from RedHat 8 to Centos 3.x- Upgrade from Centos 3.x to Centos 4.x- LD_ASSUME_KERNEL=2.4.1 - ulimit -n 65535- Upgrade from asterisk 1.0.7 to 1.0.12Currenly we can not easily migrate from asterisk-1.0.x to 1.2.xAny ideas?. Could be Debian a solution?Thank you.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retain call control: Avoid letting call get intocellular voicemail
You can create a macro that the person that is called has to press a key to take the call. If no key is pressed then you can send them to a menu where they can press one to leave a message or two to go back to the menu. On 11/14/06, joe a. <[EMAIL PROTECTED]> wrote: Did not know how to make up a subject line for this.I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible?joe a.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with FXS ports of TDM400P
Was the phone off hook when the card was started ? Also as some one mentioned above is there power being provided to the card ? Hi, I've not yet used a TDM400, only a 2400. Silly question first, are you connecting the power cable? I don't know what happens if you leave it off. I found that if I have the zaptel & asterisk services enabled, the card/drivers do not initialise correctly. To get my system working, I ended up with the services stopped. I added lines in rc.local to remove (rmmod) the card driver & zaptel modules, then start zaptel, a delay while other things are started, then start asterisk. I'm running Centos 4.4 so the exact commands may be different - eg. I use 'service zaptel start' to start zaptel, which then loads all the appropriate modules. This sequence seems to give an absolutely reliable startup (so far...) The other minor problem I had was getting the settings mixed up in /etc/zaptel.conf I initially missed the bit of the docs that said to use fxsks for the FXO modules connected to outside lines & use fxoks for the FXS modules on local extensions. Hope this is some help, Robert Jenkins. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo Felisberto Sent: 14 November 2006 21:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with FXS ports of TDM400P Gustavo Felisberto wrote: > I just received two TDM400P cards, but I'm having problems with them. > > The full info is at: > http://pastebin.com/824079 > > Extra, I'm using : > > libpri-1.2.3 > zaptel-1.2.10 > > On a x86 stable Gentoo box. > Kernel: 2.6.17 > gcc-4.1.1, glibc-2.4-r4 > > Is that an hardware problem? Should I try the other card? > I tried the other card and the problem is still there. REALY NEED HELP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Why only one out of many IP Phones re-registering every one minute
Now I know that this has something to do with the NAT for sure. But why and how, I don't know. I changed DHCP settings on my router to expire lease time in 1 hour. Then I could see on Asterisk CLI phones getting registered every hour. Then I changed it back to 7 days. But still some phones kept on re-registering every hour. The register expire setting on the phones themselves doesn't causing this message to display on Asterisk CLI, doesn't matter how frequently they are re-registering on Asterisk. I wonder how is this handled by the router or NAT device. On 11/3/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: On one of my servers I have many IP phones connected, locally and remotely, and all of them have register expiry set for 1 min. But on Asterisk CLI I see only one of them, extension 502, re-registering every single minute. None of the others re-register or info doesn't come on the CLI screen. This one phone is remotely connected. What is the reason for this. And also everytime it re-register, the port on which it is registered is changed. -- Zeeshan A Zakaria -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Grandstream TFTP system wide settings
Now I have answer to my own question, i.e. No, they don't. Grandstream Phones unfortunately are not very advanced in remote provisioning system, and they don't have one single file serving the whole installation, instead every phone needs its own configuration file. Then this file has to be converted to its binary format as well using the utility from their website. I recently installed Aastra and I was very happy with their much advanced remote provisioning system. It could also read simple text format files and mass deployment was much easier to controlBut what really impressed me later was Linksys SPA devices. I haven't tried Polycom and Snom yet, but I don't think any other phone can come any way near to the remote provisioning and control system of Linksys. It is highly advanced and very well done system. Becasue it was originally Sipura, which was intended for large scale residential deployments, so they had focused a lot on remote provisioning and device control featurs. Other than its binary format, it can read text, gunzip, xml and some encrypted formats. It has very good system of macros where the administrator doesn't have to type in MAC addresses. It also has very good HTTPS secure provisioning mechanism. And also a way for remote warm and cold reboot using SIP headers. In short words, it is just great and in future I'll prefer Linksys when it'll come to mass deployment. It has already made my life easier dealing with a client with remote extensions. And yes, SPA942 is an excellent phone too. On 11/7/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:Hi, Aastra IP Phones have two configuration files on TFTP, aastra.cfg and .cfg. Both are in text format, which makes editing easy. And aastra.cfg has system wide settings and .cfg has settings for each indivifual phones. This makes it really easy to change the global parameters system wide by changing only one aastra.cfg file. On the other hand, as I could understand, for Grandstream TFTP setup, each phones needs a separate file, which has to be edited and then converted to its own format usint ./encode.sh. There is no such file which would carry global settings for all the phones on a system. Changing 10s of configuration files for one small little thing, like daylight saving = 0, and then converting all of them to its own format is not a good way of dealing with many phones. Is there a quicker way to change settings for all Grandstream phones, is there any one file which can act as a global configuration file without changing each phones phone specific settings? And can't it be simply done by text editing, without the need to convert each file to cfg format? -- Zeeshan A Zakaria ___ --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] Caller ID in Sweden not working and looking for and voices
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: > > Hi! > I am getting inbound caller ID fine bout not out. > I am in Sweden and suing Rixtelcom /POrt80 as provider. > anyone knowing what is wrong? Assuming that is a SIP provider, it is not your job to set the callerid but the provider's - their interfacing to the regular landline network is responsible. There are providers that never send callerid, some send always (united, gmx in Germany) and some allow the user to set in his preferences wether he wants to send callerid (sipgate.de). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. Sorry if I have missed a previous answer on the question.RegardsMattias-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID in Sweden not working and looking for and voices
Hi!I am getting inbound caller ID fine bout not out.I am in Sweden and suing Rixtelcom /POrt80 as provider.anyone knowing what is wrong?Also is anyone knowing about Swedish voices to trixbox/Asterisk? I have male now and am looking fro female voices. RegardsMattias-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ --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] Re: Is asterisk able to integrate with MS SQL
Thanks, will do more research on that part. By the way, Im trying to do IVR where caller enter the pin the retrieve some information out of the MS SQL. I am wondering, what is the constraints or how to go about it. As per said MS SQL is about CDR. Now like i want to match and retrieve data out of the DB through IVR. Any guidance? Thanks. On 11/14/06, Vicky <[EMAIL PROTECTED]> wrote: oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield < [EMAIL PROTECTED] > wrote: In article <[EMAIL PROTECTED] >,Sharon Lim < [EMAIL PROTECTED]> wrote:>> Hi there,>> I am looking around, is there anyone did any integration asterisk talk to /> connect to MS SQL?Look for the package FreeTDS and install it. Then build Asterisk and it will include the TDS driver that can log CDRs to MS SQL.Alternatively, install ODBC drivers for MS SQL and then use Asterisk's ODBCfunctions.CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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] In the beginning-The first question.
James R. Stevens wrote: When you say DUAL T1 card from Digiim. Are you thinking One T for voice coming in the other T going to the remote office(s)? Why Dual T1 card? 1 for the direct link to the PSTN and the other to the Adit Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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] In the beginning-The first question.
When you say DUAL T1 card from Digiim. Are you thinking One T for voice coming in the other T going to the remote office(s)? Why Dual T1 card? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, November 14, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] In the beginning-The first question. James R. Stevens wrote: > > Have we enough info to ask: > 1) 1 server or several? > 1 for each location, if the T1 goes down, they still will have a phone system. Each Asterisk system can trunk to each other via IAX. > 2) Channel bank or not? > If you want to supply dial tone (Modems/Faxes/etc) you'll need a channel bank. > 3) Type of card for the server? > Either Digim or Sangoma Dual T1 card. Depending on your budget, you may want to look into their hardware base echo cancellers > 4) Am I having delusions of grandeur? > No, this is very doable. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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 -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ --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] Problem with FXS ports of TDM400P
Hi, I've not yet used a TDM400, only a 2400. Silly question first, are you connecting the power cable? I don't know what happens if you leave it off. I found that if I have the zaptel & asterisk services enabled, the card/drivers do not initialise correctly. To get my system working, I ended up with the services stopped. I added lines in rc.local to remove (rmmod) the card driver & zaptel modules, then start zaptel, a delay while other things are started, then start asterisk. I'm running Centos 4.4 so the exact commands may be different - eg. I use 'service zaptel start' to start zaptel, which then loads all the appropriate modules. This sequence seems to give an absolutely reliable startup (so far...) The other minor problem I had was getting the settings mixed up in /etc/zaptel.conf I initially missed the bit of the docs that said to use fxsks for the FXO modules connected to outside lines & use fxoks for the FXS modules on local extensions. Hope this is some help, Robert Jenkins. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Gustavo Felisberto > Sent: 14 November 2006 21:43 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Problem with FXS ports of TDM400P > > Gustavo Felisberto wrote: > > I just received two TDM400P cards, but I'm having problems > with them. > > > > The full info is at: > > http://pastebin.com/824079 > > > > Extra, I'm using : > > > > libpri-1.2.3 > > zaptel-1.2.10 > > > > On a x86 stable Gentoo box. > > Kernel: 2.6.17 > > gcc-4.1.1, glibc-2.4-r4 > > > > Is that an hardware problem? Should I try the other card? > > > > I tried the other card and the problem is still there. REALY > NEED HELP > > -- > Gustavo Felisberto > (HumpBack) > Web: http://dev.gentoo.org/~humpback > Blog: http://blog.felisberto.net/ > > It's most certainly GNU/Linux, not Linux. Read more at > http://www.gnu.org/gnu/why-gnu-linux.html . > - > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice mail transfer between 2 asterisk servers
Hi,I have 2 simple asterisk servers linked over IAX. I want to know if it will be possible to transfer my voice mail messages in mailbox of SIP_PHONE1 on Atserisk1 to mailbox of SIP_PHONE2 on Asterisk2.SIP_PHONE1 <-->Asterisk1 <---IAX2--> Asterisk2 <--->SIP_PHONE2 Asterisk1: v1.2.10Asterisk2: v1.2.8Thanks.Naija Man ___ --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] Problem with FXS ports of TDM400P
Roger Gulbranson wrote: > On Tue, 2006-11-14 at 17:07 -0500, Roger Gulbranson wrote: >> On Tue, 2006-11-14 at 21:42 +, Gustavo Felisberto wrote: >>> Gustavo Felisberto wrote: I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? >>> I tried the other card and the problem is still there. REALY NEED HELP >> What happens if you add: >> >> fxoks=1,2 >> >> into /etc/zaptel.conf? > > Also, what does: > > ztcfg -vv > > tell you? > > > > ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. But if i try to use the channel 3 and 4 in zapata.conf all works ok. If i add the 1 and 2 channel I get: == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 3, FXS Kewlstart signalling Nov 14 22:38:56 WARNING[5751]: chan_zap.c:1097 zt_open: Unable to specify channel 1: No such device Nov 14 22:38:56 ERROR[5751]: chan_zap.c:7237 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Nov 14 22:38:56 ERROR[5751]: chan_zap.c:12000 setup_zap: Unable to register channel '1' Nov 14 22:38:56 WARNING[5751]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 Nov 14 22:38:56 WARNING[5751]: loader.c:554 load_modules: Loading module chan_zap.so failed! -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
Hi thereI am trying to change the rtp packet size of my Cisco 7940 from 10ms to 20ms. Does anyone know how I can do this.Codec: ULAWSIP firmware: 8.2Bootload ID: PC03A300Thanks.Naija Man ___ --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] DUNDi Asterisk Cluster
On Tue, 2006-11-14 at 13:09 -0700, David Thomas wrote: > We use only IP connections to our asterisk boxes. Given this our > origination/termination providers > usually send/receive traffic to/from our network on a single IP or > limited number of IPs. > > In a DUNDi Asterisk Cluster, would each of the boxes need to be able > to connect to our origination/termination providers directly, or would > we need to setup a common gateway box to forward calls to/from our > providers? > > How is this type of routing best handled in an all IP environment? > > Thanks, > David > ___ > --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 > I would set up a separate set of boxes that can directly connect to the providers. That gives you separation of services, and if one of those boxes goes down, you'll have enough saturation to not have to worry about it. Having the main boxes do the connections to the providers adds a level of complexity when you need to bring a new system online. Just my 2 cents :) -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: "Username/auth name mismatch" + SIP phone can't connect?
Hello,
Anselm Martin Hoffmeister > Try adding username=200 which fixed things for
me. Alternatively, Try using a username that does NOT begin with a digit -
I saw a flaky softphone some time ago that would screw completely with a
numeric username.
Dovid B >The error you are getting is that asterisk has recieved the wrong
user name and or pass and is there for rejecting your registration. Your
sip.conf seems to be fine (although you may want to add dtmf and codec
settings. Test the same settings that you have now with a softphone and
see if you recieve the same errors or not.
Since the SJPhone could register OK (although sip debug showed some 401 at
some point: maybe SJPhone supports some features that Asterisk doesn't, or
at least are not supported by default), I figured it had something to do
with the GrandStream phone : I had forgotten to turn off its use of NAT
(STUN) :-/ This combined with a basic dial plan solved the issue.
For those interested, here's my basis sip.conf:
-- sip.conf ---
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
[200]
;username=200
type=friend
secret=test
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
[201]
;username=201
type=friend
secret=test
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
;host=192.168.0.234
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
-- sip.conf ---
... and the basic extensions.conf:
-- extensions.conf ---
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
[internal]
;BAD exten => ${EXTEN},1,Dial(SIP/${EXTEN})
exten => 200,1,Dial(SIP/200)
exten => 201,1,Dial(SIP/201)
exten => 202,1,Dial(SIP/202)
-- extensions.conf ---
I'm pretty sure there's a way to simplify the above, but as you can see, my
first attempt made Asterisk barf ;-)
Thanks guys for your help
Fred.
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Re: [asterisk-users] Load balance Asterisk servers?
On Tue, 2006-11-14 at 12:00 -0700, David Thomas wrote: > On 11/14/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: > > Incorrect :) IAX2 most definitely does support regcontext. > > > > Also, I think what he means is the phone specific information must be > > exactly the same from system to system or the failover won't be as > > seamless as you expect. A lot of phones support some sort of SRV > > records, so in the event of a failure, the phones will automatically > > find the next available server. The other option there is to set up an > > HA environment so the failover is even transparent to the phones, they > > just start talking to the new IP address immediately. > > > > Another thought, in any failover situation, if you have any sort of > > automated failover, you must make sure phones that need specific > > features fail to the same server (i.e. hinting and such) as those > > features don't work cross server. > > > > Aaron > > > > On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: > > > On 11/14/06, Stelios Koroneos <[EMAIL PROTECTED]> wrote: > > > > JR Richardson gave a very nice presentation at Astricon on how to do > > > > that with DUNDI > > > > > > As I understand it JR Richardson's DUNDi solution does not support > > > IAX. It uses regcontex which I believe is only available with SIP. > > > (please correct me if I'm wrong) > > > > > > Also JR notes that... > > > > > > "Associated SIP Users, business customers, require same registration > > > and failover to the same servers" so unless this is for a residential > > > setup, it may not be of much use to you. Nevertheless it is great > > > documentation, and may get you further than you are now. > > > > > > regards, > > > David > > > ___ > > > --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 > > > > -- > > Aaron Daniel > > Senior Voice Analyst > > Sam Houston State University > > [EMAIL PROTECTED] > > (936) 294-4198 > > > > ___ > > --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 > > > > There I go spreading mis-information again. :) > Thanks Aaron for clearing up the IAX2 regcontext question. That is good to > know. > > In the DUNDi scenario, are there any other features that would be > affected or become unavailable in the event of a failure? > > It seems like normal call processing would continue once the client > re-registered to a different box, but I haven't tried it yet. > > I assume one could use DNS-Round-Robin to load balance registrations > between the boxes in the cluster, then pull the failed box out of DNS > to prevent registration attempts while the box is dead. Is there a > better way to do this ??? > > David > ___ > --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 The "best practice" for the failover situation will depend on the phone you're using, and what kind of failover you're looking for. Most *real* phones support SRV records which will contain the list of servers that your phone should talk to. If the phone sees one server down, it automatically jumps to the next available one. You are correct however, that call processing would still work as long as you have the servers configured correctly :) Also, if you configure the phone correctly, you should theoretically be able to not lose any calls in case of failure as well, however you would lose call details records for that call. Douglas Garstang will tell you round-robin won't work, I haven't tested it. Something about random packets going to round-robined server addresses :) The other thing to consider is actual failover ip addresses, where one computer automatically assumes another computer's ip address in the event of failure. The phones would automatically start talking to the new system immediately. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unable to get channel lock BAD BAD BAD
I am seeing the following in my log file (standard trixbox install).
One seems to be complaining about an error in the dialplan but it
won't tell me what file or what line. The other (maybe related) is
complaining about a channel lock.
How to do go about trying to figure out what the problem is and how to solve it?
---Logfile
Nov 14 07:20:19 WARNING[14067] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL
or TOK_LP or TOKEN; Input:
"" != "
^
Nov 14 07:20:20 WARNING[14067] ast_expr2.fl: If you have questions,
please refer to doc/README.variables in the asterisk source.
Nov 14 07:20:44 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:44 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:44 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
Nov 14 07:20:45 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:45 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:45 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
Nov 14 07:20:46 ERROR[24091] chan_sip.c: We could NOT get the channel
lock for SIP/101-082695f0!
Nov 14 07:20:46 ERROR[24091] chan_sip.c: SIP MESSAGE JUST IGNORED: ACK
Nov 14 07:20:46 ERROR[24091] chan_sip.c: BAD! BAD! BAD!
--LInes in the dialplan with != in them--
asterisk -r -x 'show dialplan' | grep !=
's' =>1. GotoIf($["${REALCALLERIDNUM:1:2}" != ""]?start)
[pbx_config]
2. GotoIf($["${REALCALLERIDNUM:1:2}" != ""]?start)
[pbx_config]
2.
GotoIf($["${CALLERID(name):0:${LEN(${RGPREFIX})}}" !=
"${RGPREFIX}"]?4:3) [pbx_config]
2.
Set(VMGAIN=${IF($["foo${VM_GAIN}"!="foo"]?"g(${VM_GAIN})":"")})
[pbx_config]
'docfu' =>1.
Set(RTCFU=${IF($["${VMBOX}"!="novm"]?${RINGTIMER}:"")}) [pbx_config]
6. Set(RT=${IF($[$["${VMBOX}"!="novm"] |
$["foo${CFUEXT}"!="foo"]]?${RINGTIMER}:"")}) [pbx_config]
10. GosubIf($[$["${DIALSTATUS}"="NOANSWER"] &
$["foo${CFUEXT}"!="foo"]]?docfu|1) [pbx_config]
31. GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$["${RingGroupMethod}" = "hunt" ]]?32:35 ) [pbx_config]
35. GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$["${RingGroupMethod}" = "memoryhunt" ]]?36:50 ) [pbx_config]
2.
GotoIf($["${CALLERID(name):0:${LEN(${RGPREFIX})}}" !=
"${RGPREFIX}]"?NEWPREFIX) [pbx_config]
2.
GotoIf($["${CALLERID(name):0:${LEN(${RGPREFIX})}}" !=
"${RGPREFIX}]"?NEWPREFIX) [pbx_config]
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Re: [asterisk-users] Problem with FXS ports of TDM400P
Gustavo Felisberto wrote: > I just received two TDM400P cards, but I'm having problems with them. > > The full info is at: > http://pastebin.com/824079 > > Extra, I'm using : > > libpri-1.2.3 > zaptel-1.2.10 > > On a x86 stable Gentoo box. > Kernel: 2.6.17 > gcc-4.1.1, glibc-2.4-r4 > > Is that an hardware problem? Should I try the other card? > I tried the other card and the problem is still there. REALY NEED HELP -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --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] OutCall Release
Tzafrir Cohen wrote: > On Tue, Nov 14, 2006 at 03:41:08PM +0100, Stephen Wingfield wrote: >> LONDON, UK (14th November 2006) - Bicom Systems announced today it >> has released its first freeware software to the "Asterisk >> Community", OutCall. Tzarif, Thanks for your contribution in clarification of outCALL . > > To avoid any confusion: "free" here means a limited license for "one > copy per user". As matter of fact we do not limit how many copies one can use. Knock your self out If you wish :) > > And for those who are wondering what this OutCall is (as this press > release tell you nothing), it seems to be some sort of MS-Outlook > integration designed for either Bicom's prorietary PBXWare, or for a > standard Asterisk installation. Correct it works with either PBXware or vanilla asterisk. We had a great success with outCALL deployments. It is not our main line of product, we are not setup to collect 30-50$ per copy so what a heck, let asterisk community Enjoy it. Have a fun :) Senad ___ --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] In the beginning-The first question.
James R. Stevens wrote: Have we enough info to ask: 1) 1 server or several? 1 for each location, if the T1 goes down, they still will have a phone system. Each Asterisk system can trunk to each other via IAX. 2) Channel bank or not? If you want to supply dial tone (Modems/Faxes/etc) you'll need a channel bank. 3) Type of card for the server? Either Digim or Sangoma Dual T1 card. Depending on your budget, you may want to look into their hardware base echo cancellers 4) Am I having delusions of grandeur? No, this is very doable. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --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: Problem found Re: [asterisk-users] Headaches with Video over SIP
On Tue, 2006-11-14 at 02:10 -0800, Steve Langstaff wrote: > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Peter Howard > > Sent: 14 November 2006 00:38 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Problem found Re: [asterisk-users] Headaches with > > Video over SIP > > > Found the problem. Some other people on the list might be > > interested in the problem. > > > > Looking at the initial INVITE call from station 1 to asterisk > > and from asterisk to station 2, you see the following discrepancy: > > > > >From Polycom to asterisk: > > a=rtpmap:109 H264/9 > > a=rtpmap:96 H263-1998/9 > > > > >From asterisk to other Polycom: > > a=rtpmap:103 h263-1998/9 > > a=rtpmap:99 H264/9 > > > > The latter numbers match the numbers in rtp.c, which clearly > > don't agree with the Polycom's opinion on life. However the > > other Polycom seems to accept asterisk's version, while the > > first Polycom is still working on it's version. > > > > I modified the initialisation of static_RTP_PT in rtp.c to > > match the Polycom values and, hey presto! I have video > > working between the two stations. > > > > However, this doesn't look like a "proper" solution, if there > > are multiple opinions out there of what h263p and h264 (at > > least) should be mapped to. There's also the array > > current_RTP_PT in rtp.c What is that used for? I would have > > thought that either: > > 1) Asterisk should tell _both_ ends what mapping to use, > > 2) Asterisk should update it's own mapping based on what it's > > told (though this would have to be on a call-by-call basis) > > #2 is true (at least for audio, so I guess for video as well). > Or should be. > Codec identifiers >= 96 refer to dynamic payload types. > Thanks, that's worth knowing. > They have to be negotiated on each SDP offer/answer exchange. > > So for the Polycom->Asterisk traffic, Asterisk should parse the SDP and > say to itself "Hey, the caller wants me to send it H264 marked with > payload type 109, and/or H263-1998 marked with payload type 96." and > adapt it's outgoing payload type marking accordingly. > "should parse the SDP". It's not at 1.4.0-beta3 (or, seemingly, earlier versions). Should I submit a bug report for this? -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] In the beginning-The first question.
List, Im a Cisco certified Network guy with little telecom experience (BRI/PRI at the time) so please forgive my terminology. I am showing interest after the Network World SHSU October 4 article. We have 3 offices (Hub-Spoke T1 Frame relay to the remote offices(Data & voice on separate T)). Each office currently does their own thing for telecom. Our Main(HUB) office currently has 14 channels of T1 into an ADIT 600 punched down to the DEMARC. Our Panasonic (72 port) VB-43050 DBS picks up from the DEMARC and spits out 4 lines for our VM server. My goal is described below, the question is how to make Asterisk do it. Consolidate telecom services of the other two offices into our HUB office. Try (Hard) to keep some of the current phones (Panasonic-Digital_ Not a high priority). The Adit 600 is property of our Provider so currently we would have to take it from the other side. OR Should the T1 go straight into the Asterisk server? My Ideal: A full 24 channel T1 into an Asterisk server on its own VLAN routed to the remote offices over a separate PVC than the data. The remote offices would A) use SIP phones B) terminate somewhere allowing them to use the current phones. The smallest office has 19 stations. The largest has 25. We must have overhead paging Each office is satisfied having 7 concurrent calls MAX. Have we enough info to ask: 1) 1 server or several? 2) Channel bank or not? 3) Type of card for the server? 4) Am I having delusions of grandeur? We can talk with anyone off list or On giving much more details of our project thoughts. Surely someone (Many of you) are in the same boat. This message if the first in an attempt to gather information to document my 'Proof of Concept' to the powers that be. Thanks everyone for your reply's -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ --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] Survey: In what ways do you use Asterisk at your house?
At 02:08 AM 11/14/2006, you wrote: From memory... the strike was around $95, the relay board between $25 and $50, and the power supply was only a few dollars, so you could do it all for under $200. Do be careful with these. I was installing one and discovered that I could open it by the proper application of force. Must have had a weak spring or something as I could bounce the door and it would pop in a few seconds. They're not really intended to keep people out of your house, more to keep honest people from going through guarded doors. Ira ___ --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] OutCall Release
On Tue, Nov 14, 2006 at 03:41:08PM +0100, Stephen Wingfield wrote: > LONDON, UK (14th November 2006) - Bicom Systems announced today it has > released its first freeware software to the "Asterisk Community", OutCall. To avoid any confusion: "free" here means a limited license for "one copy per user". And for those who are wondering what this OutCall is (as this press release tell you nothing), it seems to be some sort of MS-Outlook integration designed for either Bicom's prorietary PBXWare, or for a standard Asterisk installation. (BTW: next time you want to start a new thread in the list, please don't reply to an existing message.) > > This is to be the first of similar releases of proprietary tools that can > assist users with getting the most out of Asterisk and will also be > released as freeware. > > As part of Bicom Systems wish to support the Community the Company will > also be asking if members could make donations through PayPal to > [EMAIL PROTECTED] as they see appropriate for any benefit they > receive from OutCall. These donations will be used exclusively to encourage > the development of Asterisk through the offering of Bonuses or similar > rewards. > > For full details on Bicom Systems products please visit > http://www.bicomsystems.com/products/C/P/319/288/. To download a copy of > OutCall, please visit www.bicomsystems.com Documentation is available at > www.bicomsystems.com/docs/outcall/ . > > For more information, please contact: > Stephen Wingfield > 44-20-7043-3489 > steve [at}bicomsystems [dot}com -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi Asterisk Cluster
We use only IP connections to our asterisk boxes. Given this our origination/termination providers usually send/receive traffic to/from our network on a single IP or limited number of IPs. In a DUNDi Asterisk Cluster, would each of the boxes need to be able to connect to our origination/termination providers directly, or would we need to setup a common gateway box to forward calls to/from our providers? How is this type of routing best handled in an all IP environment? Thanks, David ___ --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] Polycom - how to 'buddy watch' trunks?
Have you tried setting up a hint for a ZAP channel? exten => foo,hint,ZAP/bar Then make a directory entry for foo in your Polycom directory for foo - just as you would if the hint was for a SIP channel. CP On Nov 14, 2006, at 4:26 AM, Robert Jenkins wrote: Hi, I've recently got some Polycom 501 & 601 phones. I have buddy watch working & showing the status of users listed in the directory. I would like to also have the status of the trunks (ZAP via TDM2400E & SIP) on the IP601 Sidecar display, but I cannot so far find any info on this? Thanks, Robert. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing from "Placed Calls" on Polycom IP501doesn't always work
Hi James, We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM. Did you come up with any reason/fix for this? CP On Nov 13, 2006, at 11:00 PM, James Andrewartha wrote: Anthony Rodgers wrote: > Greetings, > > Has anyone noticed that attempting to place a call from the "Placed > Calls" list on a Polycom IP501 by pressing the 'Dial' softkey sometimes > simply returns the phone to the idle screen? It is not related to the > number being dialed, as we have observed two entries for the same > number, one of which worked and the other didn't. > > We've experimented with calls that weren't answered at all, calls that > were terminated by the caller and calls terminated by the recipient with > no discernible pattern. Yes, I've seen it. We're running 1.6.6, what firmware version do you have? -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retain call control: Avoid letting call get into cellular voicemail (was: Dialplan options)
Am Dienstag, den 14.11.2006, 12:28 -0500 schrieb joe a.: > Did not know how to make up a subject line for this. > > I have a dial plan that allows a caller can try my cell phone. And that's > fine. If the call cannot be made, it sends caller back to voice menu. > > However, I'd like a way for the caller to elect to go back to the voice menu, > if they end up getting the cell phone voice mail. Is that possible? See the Dial command information at http://www.voip-info.org/wiki/view/Asterisk+Cmd+Dial If you specify the option "H", your user can cancel the connection to the outward cellphone by pressing "*". You could send him back to your IVR, or whatever. Of course they need to know about that. In your situation, I would set the Dial() timeout to a number that is slightly below the voicemail activation timeout (20 seconds on non-answer or whatever). This avoids them hitting the "expensive" voicemail of the cellular, but you could use the DIALSTATUS information to filter them into your own voicemail IF the mobile was not answered within the predefined time. HTH Anselm ___ --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] 900 rules
Look here for details on the North American Numbering Plan: http://www.nanpa.com/reports/reports_npa.html The report named "Non-Geographic NPAs In Service" lists the Toll Free and Premium assignments. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton Sent: Tuesday, November 14, 2006 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 900 rules I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] config template for Grandstreams
It appears to be up there now. From the header: ## Configuration template for GXP-2000 firmware version 1.1.1.14 Bob... On Tue, 2006-11-14 at 13:45 -0500, Todd- Asterisk wrote: > Thanks- they did respond. I got a new template, but was asked to not > share it for now - it'll be on their website in a few days pending > committee approval > thanks > Todd > > On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote: > > > On Fri, 10 Nov 2006, Todd- Asterisk wrote: > > > >> I'm preparing to deploy a small number of Grandstream BT101's and > >> GXP2000's to a remote location (which I won't have access to). I'd > >> like to have them pull a config file from my server - I'm almost > >> there... > >> > >> The phones are looking for the config file on my webserver which is > >> good. I need to generate that file however. I see a tool on the GS > >> website to generate the config file from a template, but the > >> templates posted on their website are for an old version of the phone > >> firmware. Anyone have a tool or access to templates for the latest > >> firmware versions? > > > > Email their technical support. I did this a few days ago for the > > latest > > one for the GPX2000 and they emailled it back the next day. > > > > Gordon > > ___ > > --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
Re: [asterisk-users] Load balance Asterisk servers?
On 11/14/06, Aaron Daniel <[EMAIL PROTECTED]> wrote: Incorrect :) IAX2 most definitely does support regcontext. Also, I think what he means is the phone specific information must be exactly the same from system to system or the failover won't be as seamless as you expect. A lot of phones support some sort of SRV records, so in the event of a failure, the phones will automatically find the next available server. The other option there is to set up an HA environment so the failover is even transparent to the phones, they just start talking to the new IP address immediately. Another thought, in any failover situation, if you have any sort of automated failover, you must make sure phones that need specific features fail to the same server (i.e. hinting and such) as those features don't work cross server. Aaron On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: > On 11/14/06, Stelios Koroneos <[EMAIL PROTECTED]> wrote: > > JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI > > As I understand it JR Richardson's DUNDi solution does not support > IAX. It uses regcontex which I believe is only available with SIP. > (please correct me if I'm wrong) > > Also JR notes that... > > "Associated SIP Users, business customers, require same registration > and failover to the same servers" so unless this is for a residential > setup, it may not be of much use to you. Nevertheless it is great > documentation, and may get you further than you are now. > > regards, > David > ___ > --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 There I go spreading mis-information again. :) Thanks Aaron for clearing up the IAX2 regcontext question. That is good to know. In the DUNDi scenario, are there any other features that would be affected or become unavailable in the event of a failure? It seems like normal call processing would continue once the client re-registered to a different box, but I haven't tried it yet. I assume one could use DNS-Round-Robin to load balance registrations between the boxes in the cluster, then pull the failed box out of DNS to prevent registration attempts while the box is dead. Is there a better way to do this ??? David ___ --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] config template for Grandstreams
I see the Grandstream website now has the new config templates posted with all the happy P commands... http://grandstream.com/y-configurationtool.htm Todd ___ --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] Newbie Questions . . .
By the time you purchase PCI cards for you extensions (FSO ports)you would be better off purchasing SIP phones like Grandstream GXP 2000 this will give you a fully featured PBX IP phone for about the same cost or less than FSO ports. Asterisk will have no problem running 25 or more SIP phones Personally I would reduce the incoming analog lines to 4 (FXO) ports and add some DID lines. This way you will only have to buy one PCI board with 4 FXO ports Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada > Maybe you should try this > http://www.digium.com/en/products/hardware/aadk.php . > Is very heavy loaded if 9PCI cards at a server. But is possible but not > encourge. Maybe you can consider to have digital extension with IP phone. > THis is my opinion. > > :-) good luck > > On 11/14/06, Jason Flatt <[EMAIL PROTECTED]> wrote: >> >> Hello all. >> >> My company currently has an older Executone PBX system that we are >> outgrowing. >> Rather than wait until the last minute to make a hasty decision, I >> thought >> it >> would be a good idea to do some research and compare options first. My >> expertise is in computers and networking, and telephony systems are >> mostly >> foreign to me. >> >> What we currently have are 5 incoming POTS lines and 25 stations and are >> wanting to add 1 or 2 more stations. I think we might have added at >> least >> one more incoming line, except that the phones we have only support 5 >> lines >> (so I'm told). Our PBX system has room for 5 more stations, then it's >> time >> to buy a new one. >> >> I'm assuming I need to add some hardware in order to make Asterisk work >> with >> our existing setup, but I'm not entirely sure what. Based on the >> reading >> I've done so far and my limited understanding, if we wanted to use it in >> place of our existing PBX system, I would need to get an "analog >> interface >> card" (several, actually), like Digium's TDM400P, like so: >> >> 2 - Wildcard TDM04B cards for FXO and >> 7 - Wildcard TDM40B cards for FXS >> >> -or- >> >> 1 - Wildcard TDM04B card for FXO and >> 1 - Wildcard TDM22B card for FXO & FXS and >> 7 - Wildcard TDM40B cards for FXS >> >> I might as well use the top configuration for future expansion. >> >> If I am correct, that is 9 PCI cards in a PC. I don't know of any >> motherboard >> that supports that many cards, so either I'm wrong, or I'll need >> different >> cards, or I'll need to utilize 2 or more PCs in conjunction with each >> other. >> I haven't yet found any mention on the last two options, so I'm assuming >> I'm >> wrong and I need a little enlightenment. >> >> Thank you for any information that will help me better understand this. >> >> >> -- >> Jason Flatt >> Father of Six: http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; >> Travis, >> 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005) >> Linux User: http://www.sourcemage.org/ >> Drupal Fanatic: http://drupal.org/ >> ___ >> --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 >> > > > > -- > Regards, > Sharon Lim > > *Good memories are to be folded neatly and tucked away into the back > pocket > * > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] config template for Grandstreams
Thanks- they did respond. I got a new template, but was asked to not share it for now - it'll be on their website in a few days pending committee approval thanks Todd On Nov 14, 2006, at 12:50 PM, Gordon Henderson wrote: On Fri, 10 Nov 2006, Todd- Asterisk wrote: I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate the config file from a template, but the templates posted on their website are for an old version of the phone firmware. Anyone have a tool or access to templates for the latest firmware versions? Email their technical support. I did this a few days ago for the latest one for the GPX2000 and they emailled it back the next day. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Load balance Asterisk servers?
> -Original Message- > From: Aaron Daniel [mailto:[EMAIL PROTECTED] > Sent: Tuesday, November 14, 2006 9:24 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Load balance Asterisk servers? > > > Incorrect :) IAX2 most definitely does support regcontext. > > Also, I think what he means is the phone specific information must be > exactly the same from system to system or the failover won't be as > seamless as you expect. A lot of phones support some sort of SRV > records, so in the event of a failure, the phones will automatically > find the next available server. The other option there is to > set up an > HA environment so the failover is even transparent to the phones, they > just start talking to the new IP address immediately. > > Another thought, in any failover situation, if you have any sort of > automated failover, you must make sure phones that need specific > features fail to the same server (i.e. hinting and such) as those > features don't work cross server. To put things in perspective, one could consider that BLF is not a critical function, and in a failover situation, if call processing continues, but your BLF function is degraded (until the phones resubscribe), then that might be acceptable. Doug. ___ --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] 900 rules
There are many none free area codes in the 8xx space. Just google "area codes". .Brian Doug Crompton wrote: > Ok so ONLY 900 numbers are pay. > > Next question 18XX numbers. are they all toll free? Is there any > space in 8xx that is used otherwise? > > Doug > > > On Tue, 14 Nov 2006, Eric "ManxPower" Wieling wrote: > >> Doug Crompton wrote: >>> I had a 19xx rule in asterisk and realized when I was trying to dial an >>> area code 978 in MA that that was not a good idea. Is there a more defined >>> rule for 900 space of non pay vs. pay codes? > > >> _1900NXX >> _NXX976 >> ___ >> > > ___ > --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
[asterisk-users] Re: Broken Call Screening
You need to modify app_queue.c to hold off on bridging until the receiving party has accepted the call. If the receiving party rejects (hangup, digit other than '1', timeout, etc), leave or put the calling party back in at close to the same level.Justin--Date: Tue, 14 Nov 2006 10:14:04 -0500From: "Gary T. Giesen" <[EMAIL PROTECTED]>Subject: [asterisk-users] Broken Call Screening...I have a cell phone added to a queue as a local extension (member =>Local/299). I want the cell phone to be able to reject calls to thequeue without the person sitting in the queue being hung up on, etc.The way my dialplan is set up, the person hits 1 to answer the calland any other key to reject it. It works flawlessly in that regard.If it goes to the cell phone voicemail, it works great too, it timesout and rejects the call, all without the caller knowing. Where itbreaks is when the person answers the cell phone and then hangs upwithout any input or letting it time out. The music on hold is stoppedand the caller is left there with dead air. Does anyone have any ideason how to fix this or a better way to implement this?Output when the call is dropped: -- Channel 0/3, span 1 got hangup request -- User disconnected -- Stopped music on hold on Local/[EMAIL PROTECTED],2Nov 13 16:21:26 WARNING[12709]: res_features.c:1374 ast_bridge_call:Bridge failed on channels Local/[EMAIL PROTECTED],2 and Zap/3-1 -- Hungup 'Zap/3-1' -- Local/[EMAIL PROTECTED],1 answered SIP/7960A-Gary1-63f2 -- Stopped music on hold on SIP/7960A-Gary1-63f2...Regards,Gary___ --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] Dual Wan Router with Failover
We've had great results with Astrocom powerlink for load balancing outbound wan connections. == Jeronimo Romero EUS Networks Email: [EMAIL PROTECTED] Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com == > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Dean Collins > Sent: Tuesday, November 14, 2006 12:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: [EMAIL PROTECTED] > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > Thanks for the reply matthew, basically I've been looking at going to a > dmz model for a while as currently everything runs through a single > sbs2003 server and when it's churning drives doing something sometimes > audio errors occur. > > > Cheers, > > Dean > > > > -Original Message- > > From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, 14 November 2006 11:41 AM > > To: Dean Collins > > Cc: Asterisk-Users > > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > > > There are several dual-WAN routers with load balancing and > failover, > > including the Xincom Twin-WAN series that I have tested OK with SIP > (as > > NAT): http://www.xincom.com/twinwan.php . Their other products > probably > > work, too. > > > > Keep in mind that load balancing on these devices assigns each > > TCP/IP > > *connection* to its own WAN interface. So a large transfer on a single > > connection is limited by the bandwidth of whichever interface it's > > started on, even if the transfer starts slow enough to get assigned to > a > > smaller bandwidth interface, then expands to require the bandwidth > from > > the other WAN. The tech to de/multiplex streams works "well" only when > > connecting to a single router endpoint, over relatively few hops that > > can lengthen unpredictably the path some deplexed tackets travel. UDP > > works better, but it still doesn't really work that well. What works > > well is assigning different WANs to different apps' traffic, using > > multiple WANs for failover, or just accepting that these techs are > > better than nothing, and offer cheap ways to at least avoid a single > > point of failure in the WAN scheme. > > > > > > On Tue, 2006-11-14 at 08:07 -0700, > > [EMAIL PROTECTED] wrote: > > > Date: Tue, 14 Nov 2006 10:07:41 -0500 > > > From: "Dean Collins" <[EMAIL PROTECTED]> > > > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > > > Message-ID: > > > > > <[EMAIL PROTECTED]> > > > Content-Type: text/plain; charset="US-ASCII" > > > > > > Hi Jason, > > > I was looking for an external solution outside of my asterisk box so > > > that I can load balance my other website/email traffic as well. > > > > > > > > > Cheers, > > > > > > Dean > > > > > > > > > > -Original Message- > > > > From: > > > [EMAIL PROTECTED] [mailto:asterisk-users- > > > > [EMAIL PROTECTED] On Behalf Of Jason > > > > Sent: Tuesday, 14 November 2006 11:00 AM > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > If you don't mind using linux, linux can do some fairly intense > load > > > > balancing all built in. Check out the Linux Virtual Server > project. > > > As > > > > for WAN failover, if you again don't mind using linux, you can > > > script > > > a > > > > simple ping to the internet (I would ping at least 3 hosts) and if > > > that > > > > fails, fail to your second ISP. You can also do some crazy fun > > > stuff > > > > with linux advance routing and bonding. > > > > > > > > Jason > > > > The place where you made your stand never mattered, > > > > only that you were there... and still on your feet > > > > > > > > > > > > > > > > Dean Collins wrote: > > > > > > > > > > Are you looking for load balancing or failover. > > > > > > > > > > > > > > > > > > > > Also is there a cheaper way of implementing load balancing than > > > $845 > > > > > appliance? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Cheers, > > > > > > > > > > > > > > > > > > > > Dean > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > > > > [mailto:[EMAIL PROTECTED] *On Behalf Of > > > *Todd- > > > > > Asterisk > > > > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > > > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > > > > > > > > > > > > > I've been looking for this as well.. I need to support up to 20 > > > VOIP > > > > > phones over Internet as the Asterisk server is off-site. We'll > > > have > > > > > multiple cable modems or DSL routers. > > > > > > > > > > > > > > > > > > > > I found this device
RE: [asterisk-users] Dual Wan Router with Failover
Thanks for the reply matthew, basically I've been looking at going to a dmz model for a while as currently everything runs through a single sbs2003 server and when it's churning drives doing something sometimes audio errors occur. Cheers, Dean > -Original Message- > From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] > Sent: Tuesday, 14 November 2006 11:41 AM > To: Dean Collins > Cc: Asterisk-Users > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > There are several dual-WAN routers with load balancing and failover, > including the Xincom Twin-WAN series that I have tested OK with SIP (as > NAT): http://www.xincom.com/twinwan.php . Their other products probably > work, too. > > Keep in mind that load balancing on these devices assigns each > TCP/IP > *connection* to its own WAN interface. So a large transfer on a single > connection is limited by the bandwidth of whichever interface it's > started on, even if the transfer starts slow enough to get assigned to a > smaller bandwidth interface, then expands to require the bandwidth from > the other WAN. The tech to de/multiplex streams works "well" only when > connecting to a single router endpoint, over relatively few hops that > can lengthen unpredictably the path some deplexed tackets travel. UDP > works better, but it still doesn't really work that well. What works > well is assigning different WANs to different apps' traffic, using > multiple WANs for failover, or just accepting that these techs are > better than nothing, and offer cheap ways to at least avoid a single > point of failure in the WAN scheme. > > > On Tue, 2006-11-14 at 08:07 -0700, > [EMAIL PROTECTED] wrote: > > Date: Tue, 14 Nov 2006 10:07:41 -0500 > > From: "Dean Collins" <[EMAIL PROTECTED]> > > Subject: RE: [asterisk-users] Dual Wan Router with Failover > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Message-ID: > > > <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset="US-ASCII" > > > > Hi Jason, > > I was looking for an external solution outside of my asterisk box so > > that I can load balance my other website/email traffic as well. > > > > > > Cheers, > > > > Dean > > > > > > > -Original Message- > > > From: > > [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] On Behalf Of Jason > > > Sent: Tuesday, 14 November 2006 11:00 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > > > > > If you don't mind using linux, linux can do some fairly intense load > > > balancing all built in. Check out the Linux Virtual Server project. > > As > > > for WAN failover, if you again don't mind using linux, you can > > script > > a > > > simple ping to the internet (I would ping at least 3 hosts) and if > > that > > > fails, fail to your second ISP. You can also do some crazy fun > > stuff > > > with linux advance routing and bonding. > > > > > > Jason > > > The place where you made your stand never mattered, > > > only that you were there... and still on your feet > > > > > > > > > > > > Dean Collins wrote: > > > > > > > > Are you looking for load balancing or failover. > > > > > > > > > > > > > > > > Also is there a cheaper way of implementing load balancing than > > $845 > > > > appliance? > > > > > > > > > > > > > > > > > > > > > > > > Cheers, > > > > > > > > > > > > > > > > Dean > > > > > > > > > > > > > > > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] *On Behalf Of > > *Todd- > > > > Asterisk > > > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > > > > > > > > > I've been looking for this as well.. I need to support up to 20 > > VOIP > > > > phones over Internet as the Asterisk server is off-site. We'll > > have > > > > multiple cable modems or DSL routers. > > > > > > > > > > > > > > > > I found this device which looks promising - does anyone have any > > > > experience with this? > > > > > > > > http://www.peplink.com/productsLoader.php?productName=balance > > > > > > > > > > > > > > > > Todd > > > > > > > > > > > > > > > > On Nov 13, 2006, at 8:49 PM, Dovid B wrote: > > > > > > > > > > > > > > > > Hi List, > > > > > > > > Does anyone know of a good dual wan router that can handle SIP > > well > > > > and can failover between connections if there is a SIP issue on > > one > > of > > > > the lines (meaning there still is a connection however there isnt > > > > enough bandwith or sip packets arent going thru etc.) ? > > > > > > > > > > > > > > > > Thanks. > > > > > > > > > > > > > > > > Dovid > -- > > (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- aster
Re: [asterisk-users] 900 rules
On Tue, Nov 14, 2006 at 10:39:00AM -0500, Steve Sobol wrote: > On Tue, 14 Nov 2006, Doug Crompton wrote: > > Ok so ONLY 900 numbers are pay. > > > > Next question 18XX numbers. are they all toll free? Is there any > > space in 8xx that is used otherwise? > > Whoa. No, only 800, 888, 877, 866 and futuree NPA's 855, 844, 833, and 822 > are toll free. And some of those apparently *forward* to numbers which cram your bill, with questionable legality. There's no 100% solution. Cheers, -- jr 'or even a 7% one, I sometimes think' a -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --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] config template for Grandstreams
On Fri, 10 Nov 2006, Todd- Asterisk wrote: > I'm preparing to deploy a small number of Grandstream BT101's and > GXP2000's to a remote location (which I won't have access to). I'd > like to have them pull a config file from my server - I'm almost > there... > > The phones are looking for the config file on my webserver which is > good. I need to generate that file however. I see a tool on the GS > website to generate the config file from a template, but the > templates posted on their website are for an old version of the phone > firmware. Anyone have a tool or access to templates for the latest > firmware versions? Email their technical support. I did this a few days ago for the latest one for the GPX2000 and they emailled it back the next day. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk sip doesn't see other asterisk-sip
Hello, Here is our setup: asterisk-A <--LAN--> nat-router <--Internet--> asterisk-B A and B have appropriate friend entries in their sip.conf with a qualify=yes. The router forwards anything on sip,iax and sip/rtp ports to A. The problem: SIP/A remains UNREACHABLE for SIP/B, however A sees B. No problem with iax2. What did I miss in my configuration? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Retain call control: Avoid letting call get into cellular voicemail
Try this subject line if you will. On 11/14/06, joe a. <[EMAIL PROTECTED]> wrote: Did not know how to make up a subject line for this.I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible?joe a.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
[asterisk-users] Dialplan options
Did not know how to make up a subject line for this. I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible? joe a. ___ --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] Load balance Asterisk servers?
Incorrect :) IAX2 most definitely does support regcontext. Also, I think what he means is the phone specific information must be exactly the same from system to system or the failover won't be as seamless as you expect. A lot of phones support some sort of SRV records, so in the event of a failure, the phones will automatically find the next available server. The other option there is to set up an HA environment so the failover is even transparent to the phones, they just start talking to the new IP address immediately. Another thought, in any failover situation, if you have any sort of automated failover, you must make sure phones that need specific features fail to the same server (i.e. hinting and such) as those features don't work cross server. Aaron On Tue, 2006-11-14 at 08:16 -0700, David Thomas wrote: > On 11/14/06, Stelios Koroneos <[EMAIL PROTECTED]> wrote: > > JR Richardson gave a very nice presentation at Astricon on how to do that > > with DUNDI > > As I understand it JR Richardson's DUNDi solution does not support > IAX. It uses regcontex which I believe is only available with SIP. > (please correct me if I'm wrong) > > Also JR notes that... > > "Associated SIP Users, business customers, require same registration > and failover to the same servers" so unless this is for a residential > setup, it may not be of much use to you. Nevertheless it is great > documentation, and may get you further than you are now. > > regards, > David > ___ > --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 -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with voicemail
Hi all, When I listen to my voicemail or when the caller reviews his message the volume is too high in the playback. It is allmost distorted. The other sounds of the PBX sounds great, its just the voicemail that is being plaied. Any thoughts? many thanks, Christian ___ --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] Dual Wan Router with Failover
There are several dual-WAN routers with load balancing and failover, including the Xincom Twin-WAN series that I have tested OK with SIP (as NAT): http://www.xincom.com/twinwan.php . Their other products probably work, too. Keep in mind that load balancing on these devices assigns each TCP/IP *connection* to its own WAN interface. So a large transfer on a single connection is limited by the bandwidth of whichever interface it's started on, even if the transfer starts slow enough to get assigned to a smaller bandwidth interface, then expands to require the bandwidth from the other WAN. The tech to de/multiplex streams works "well" only when connecting to a single router endpoint, over relatively few hops that can lengthen unpredictably the path some deplexed tackets travel. UDP works better, but it still doesn't really work that well. What works well is assigning different WANs to different apps' traffic, using multiple WANs for failover, or just accepting that these techs are better than nothing, and offer cheap ways to at least avoid a single point of failure in the WAN scheme. On Tue, 2006-11-14 at 08:07 -0700, [EMAIL PROTECTED] wrote: > Date: Tue, 14 Nov 2006 10:07:41 -0500 > From: "Dean Collins" <[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] Dual Wan Router with Failover > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > Hi Jason, > I was looking for an external solution outside of my asterisk box so > that I can load balance my other website/email traffic as well. > > > Cheers, > > Dean > > > > -Original Message- > > From: > [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Jason > > Sent: Tuesday, 14 November 2006 11:00 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > > > If you don't mind using linux, linux can do some fairly intense load > > balancing all built in. Check out the Linux Virtual Server project. > As > > for WAN failover, if you again don't mind using linux, you can > script > a > > simple ping to the internet (I would ping at least 3 hosts) and if > that > > fails, fail to your second ISP. You can also do some crazy fun > stuff > > with linux advance routing and bonding. > > > > Jason > > The place where you made your stand never mattered, > > only that you were there... and still on your feet > > > > > > > > Dean Collins wrote: > > > > > > Are you looking for load balancing or failover. > > > > > > > > > > > > Also is there a cheaper way of implementing load balancing than > $845 > > > appliance? > > > > > > > > > > > > > > > > > > Cheers, > > > > > > > > > > > > Dean > > > > > > > > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] *On Behalf Of > *Todd- > > > Asterisk > > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > > > > > I've been looking for this as well.. I need to support up to 20 > VOIP > > > phones over Internet as the Asterisk server is off-site. We'll > have > > > multiple cable modems or DSL routers. > > > > > > > > > > > > I found this device which looks promising - does anyone have any > > > experience with this? > > > > > > http://www.peplink.com/productsLoader.php?productName=balance > > > > > > > > > > > > Todd > > > > > > > > > > > > On Nov 13, 2006, at 8:49 PM, Dovid B wrote: > > > > > > > > > > > > Hi List, > > > > > > Does anyone know of a good dual wan router that can handle SIP > well > > > and can failover between connections if there is a SIP issue on > one > of > > > the lines (meaning there still is a connection however there isnt > > > enough bandwith or sip packets arent going thru etc.) ? > > > > > > > > > > > > Thanks. > > > > > > > > > > > > Dovid -- (C) Matthew Rubenstein ___ --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] Dual Wan Router with Failover
you could use any number of the linux firewalls out there then. I know some folks are really happy with smoothwall. I just use some hacked up iptables scripts myself. Jason The place where you made your stand never mattered, only that you were there... and still on your feet Dean Collins wrote: > Hi Jason, > I was looking for an external solution outside of my asterisk box so > that I can load balance my other website/email traffic as well. > > > Cheers, > > Dean > > > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of Jason >> Sent: Tuesday, 14 November 2006 11:00 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Dual Wan Router with Failover >> >> If you don't mind using linux, linux can do some fairly intense load >> balancing all built in. Check out the Linux Virtual Server project. >> > As > >> for WAN failover, if you again don't mind using linux, you can script >> > a > >> simple ping to the internet (I would ping at least 3 hosts) and if >> > that > >> fails, fail to your second ISP. You can also do some crazy fun stuff >> with linux advance routing and bonding. >> >> Jason >> The place where you made your stand never mattered, >> only that you were there... and still on your feet >> >> >> >> Dean Collins wrote: >> >>> Are you looking for load balancing or failover. >>> >>> >>> >>> Also is there a cheaper way of implementing load balancing than $845 >>> appliance? >>> >>> >>> >>> >>> >>> Cheers, >>> >>> >>> >>> Dean >>> >>> >>> >>> >>> > > >>> *From:* [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] *On Behalf Of >>> > *Todd- > >>> Asterisk >>> *Sent:* Tuesday, 14 November 2006 9:26 AM >>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >>> *Subject:* Re: [asterisk-users] Dual Wan Router with Failover >>> >>> >>> >>> I've been looking for this as well.. I need to support up to 20 >>> > VOIP > >>> phones over Internet as the Asterisk server is off-site. We'll have >>> multiple cable modems or DSL routers. >>> >>> >>> >>> I found this device which looks promising - does anyone have any >>> experience with this? >>> >>> http://www.peplink.com/productsLoader.php?productName=balance >>> >>> >>> >>> Todd >>> >>> >>> >>> On Nov 13, 2006, at 8:49 PM, Dovid B wrote: >>> >>> >>> >>> Hi List, >>> >>> Does anyone know of a good dual wan router that can handle SIP well >>> and can failover between connections if there is a SIP issue on one >>> > of > >>> the lines (meaning there still is a connection however there isnt >>> enough bandwith or sip packets arent going thru etc.) ? >>> >>> >>> >>> Thanks. >>> >>> >>> >>> Dovid >>> >>> >>> >>> >>> > > >>> ___ >>> --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
RE: [asterisk-users] Dual Wan Router with Failover
Sweet, now that is interesting http://hotbrick.com/produto.asp?tipo=2&codPro=22 anyone have any comments on the load balancing capability of these? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mailing List Sent: Tuesday, 14 November 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dual Wan Router with Failover http://hotbrick.com/ - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 9:38 AM Subject: RE: [asterisk-users] Dual Wan Router with Failover Are you looking for load balancing or failover. Also is there a cheaper way of implementing load balancing than $845 appliance? ___ --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] Load balance Asterisk servers?
On 11/14/06, Stelios Koroneos <[EMAIL PROTECTED]> wrote: JR Richardson gave a very nice presentation at Astricon on how to do that with DUNDI As I understand it JR Richardson's DUNDi solution does not support IAX. It uses regcontex which I believe is only available with SIP. (please correct me if I'm wrong) Also JR notes that... "Associated SIP Users, business customers, require same registration and failover to the same servers" so unless this is for a residential setup, it may not be of much use to you. Nevertheless it is great documentation, and may get you further than you are now. regards, David ___ --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] 900 rules
Not all 8XX numbers are free. 800, 888, 877, 866, ans 855 are free. I don't remember when 855 is scheduled to start being issued. Try checking your local phone book. Doug Crompton wrote: Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric "ManxPower" Wieling wrote: Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broken Call Screening
Sorry for the crosspost (this was also posted to
asterisk-at-uc-dot-org) but I haven't got a response.
I have a cell phone added to a queue as a local extension (member =>
Local/299). I want the cell phone to be able to reject calls to the
queue without the person sitting in the queue being hung up on, etc.
The way my dialplan is set up, the person hits 1 to answer the call
and any other key to reject it. It works flawlessly in that regard.
If it goes to the cell phone voicemail, it works great too, it times
out and rejects the call, all without the caller knowing. Where it
breaks is when the person answers the cell phone and then hangs up
without any input or letting it time out. The music on hold is stopped
and the caller is left there with dead air. Does anyone have any ideas
on how to fix this or a better way to implement this?
Output when the call is dropped:
-- Channel 0/3, span 1 got hangup request
-- User disconnected
-- Stopped music on hold on Local/[EMAIL PROTECTED],2
Nov 13 16:21:26 WARNING[12709]: res_features.c:1374 ast_bridge_call:
Bridge failed on channels Local/[EMAIL PROTECTED],2 and Zap/3-1
-- Hungup 'Zap/3-1'
-- Local/[EMAIL PROTECTED],1 answered SIP/7960A-Gary1-63f2
-- Stopped music on hold on SIP/7960A-Gary1-63f2
Dialplan:
exten => 299,1,Wait(0.2)
exten => 299,n,Dial(Zap/3/${CELLNUMBER}|60|gmM(screen^${SCREEN_FILE}))
[macro-screen]
exten => s,1,Wait(0.2)
exten => s,n,Answer
exten => s,n,Read(ACCEPT|${ARG1}|1)
exten => s,n,GotoIf($[${ACCEPT} = 1 ] ?yes:no)
exten => s,n(no),SetVar(MACRO_RESULT=CONTINUE)
Regards,
Gary
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Re: [asterisk-users] Dual Wan Router with Failover
http://hotbrick.com/ - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 9:38 AM Subject: RE: [asterisk-users] Dual Wan Router with Failover Are you looking for load balancing or failover. Also is there a cheaper way of implementing load balancing than $845 appliance? ___ --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] 900 rules
Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric "ManxPower" Wieling wrote: > Doug Crompton wrote: > > I had a 19xx rule in asterisk and realized when I was trying to dial an > > area code 978 in MA that that was not a good idea. Is there a more defined > > rule for 900 space of non pay vs. pay codes? > > _1900NXX > _NXX976 > ___ > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 900 rules
I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with FXS ports of TDM400P
I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ It's most certainly GNU/Linux, not Linux. Read more at http://www.gnu.org/gnu/why-gnu-linux.html . - signature.asc Description: OpenPGP digital signature ___ --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] Desktop integration
Tim Panton wrote: > On 13 Nov 2006, at 13:15, Ondrej Valousek wrote: > >> Hi Dean, >> >> I will check that site - thanks for the hint. >> The biggest problem I see with authentication and I do not think >> mexuar could help me here (and I am definitely going to pay $2000 for >> it :-) But it is another story... Hi I have not had time to read this post from beginning. However maybe a FREE or partner copy of outCALL may suit your needs. More details at: www.bicomsystems.com Senad ___ --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] Dual Wan Router with Failover
Dean, A small Linux box will make a very effective router (and firewall if required) and give load balancing/failover capabilities. I've done it in the past (many moons ago!) A link from my bookmarks: http://lartc.org/ - can be a little scary depending on your knowledge of ip routing and linux but there are plenty of examples to help! Jordan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: 14 November 2006 15:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Dual Wan Router with Failover Hi Jason, I was looking for an external solution outside of my asterisk box so that I can load balance my other website/email traffic as well. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason > Sent: Tuesday, 14 November 2006 11:00 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > If you don't mind using linux, linux can do some fairly intense load > balancing all built in. Check out the Linux Virtual Server project. As > for WAN failover, if you again don't mind using linux, you can script a > simple ping to the internet (I would ping at least 3 hosts) and if that > fails, fail to your second ISP. You can also do some crazy fun stuff > with linux advance routing and bonding. > > Jason > The place where you made your stand never mattered, only that you were > there... and still on your feet > > > > Dean Collins wrote: > > > > Are you looking for load balancing or failover. > > > > > > > > Also is there a cheaper way of implementing load balancing than $845 > > appliance? > > > > > > > > > > > > Cheers, > > > > > > > > Dean > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Todd- > > Asterisk > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > I've been looking for this as well.. I need to support up to 20 VOIP > > phones over Internet as the Asterisk server is off-site. We'll have > > multiple cable modems or DSL routers. > > > > > > > > I found this device which looks promising - does anyone have any > > experience with this? > > > > http://www.peplink.com/productsLoader.php?productName=balance > > > > > > > > Todd > > > > > > > > On Nov 13, 2006, at 8:49 PM, Dovid B wrote: > > > > > > > > Hi List, > > > > Does anyone know of a good dual wan router that can handle SIP well > > and can failover between connections if there is a SIP issue on one of > > the lines (meaning there still is a connection however there isnt > > enough bandwith or sip packets arent going thru etc.) ? > > > > > > > > Thanks. > > > > > > > > Dovid > > > > > > > > > > > > ___ > > --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
Re: [asterisk-users] 900 rules
There are several Caribbean countries within the 8XX range, as well as more toll free, and regular area codes. Here's the full list at NANPA: http://www.nationalnanpa.com/nas/public/npasInServiceByNumberReport.do?method=displayNpasInServiceByNumberReport Doug Crompton wrote: Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric "ManxPower" Wieling wrote: Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ ___ --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 -- Chris Mazuc Systems Administrator DataGroup Technologies (252)329-1382 ___ --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] 900 rules
810 is an area code in Michigan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Doug Crompton Sent: Tuesday, November 14, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 900 rules Ok so ONLY 900 numbers are pay. Next question 18XX numbers. are they all toll free? Is there any space in 8xx that is used otherwise? Doug On Tue, 14 Nov 2006, Eric "ManxPower" Wieling wrote: > Doug Crompton wrote: > > I had a 19xx rule in asterisk and realized when I was trying to dial an > > area code 978 in MA that that was not a good idea. Is there a more defined > > rule for 900 space of non pay vs. pay codes? > > _1900NXX > _NXX976 > ___ > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dual Wan Router with Failover
I've been looking for this as well.. I need to support up to 20 VOIP phones over Internet as the Asterisk server is off-site. We'll have multiple cable modems or DSL routers. I found this device which looks promising - does anyone have any experience with this? http://www.peplink.com/productsLoader.php?productName=balance ToddOn Nov 13, 2006, at 8:49 PM, Dovid B wrote: Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid___ --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] Dual Wan Router with Failover
Hi Jason, I was looking for an external solution outside of my asterisk box so that I can load balance my other website/email traffic as well. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason > Sent: Tuesday, 14 November 2006 11:00 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Dual Wan Router with Failover > > If you don't mind using linux, linux can do some fairly intense load > balancing all built in. Check out the Linux Virtual Server project. As > for WAN failover, if you again don't mind using linux, you can script a > simple ping to the internet (I would ping at least 3 hosts) and if that > fails, fail to your second ISP. You can also do some crazy fun stuff > with linux advance routing and bonding. > > Jason > The place where you made your stand never mattered, > only that you were there... and still on your feet > > > > Dean Collins wrote: > > > > Are you looking for load balancing or failover. > > > > > > > > Also is there a cheaper way of implementing load balancing than $845 > > appliance? > > > > > > > > > > > > Cheers, > > > > > > > > Dean > > > > > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *Todd- > > Asterisk > > *Sent:* Tuesday, 14 November 2006 9:26 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* Re: [asterisk-users] Dual Wan Router with Failover > > > > > > > > I've been looking for this as well.. I need to support up to 20 VOIP > > phones over Internet as the Asterisk server is off-site. We'll have > > multiple cable modems or DSL routers. > > > > > > > > I found this device which looks promising - does anyone have any > > experience with this? > > > > http://www.peplink.com/productsLoader.php?productName=balance > > > > > > > > Todd > > > > > > > > On Nov 13, 2006, at 8:49 PM, Dovid B wrote: > > > > > > > > Hi List, > > > > Does anyone know of a good dual wan router that can handle SIP well > > and can failover between connections if there is a SIP issue on one of > > the lines (meaning there still is a connection however there isnt > > enough bandwith or sip packets arent going thru etc.) ? > > > > > > > > Thanks. > > > > > > > > Dovid > > > > > > > > > > > > ___ > > --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
Re: [asterisk-users] Can I disable send e-mail feature in the voicemail application?
just dont enter any email address while creating extension / mailbox ;)On 14/11/06, Ma Zhiyong <[EMAIL PROTECTED] > wrote: HI, allCan I disable send e-mail feature in the voicemail application?___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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
[asterisk-users] OutCall Release
LONDON, UK (14th November 2006) - Bicom Systems announced today it has released its first freeware software to the "Asterisk Community", OutCall. This is to be the first of similar releases of proprietary tools that can assist users with getting the most out of Asterisk and will also be released as freeware. As part of Bicom Systems wish to support the Community the Company will also be asking if members could make donations through PayPal to [EMAIL PROTECTED] as they see appropriate for any benefit they receive from OutCall. These donations will be used exclusively to encourage the development of Asterisk through the offering of Bonuses or similar rewards. For full details on Bicom Systems products please visit http://www.bicomsystems.com/products/C/P/319/288/. To download a copy of OutCall, please visit www.bicomsystems.com Documentation is available at www.bicomsystems.com/docs/outcall/ . For more information, please contact: Stephen Wingfield 44-20-7043-3489 steve [at}bicomsystems [dot}com ___ --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] 900 rules
Doug Crompton wrote: I had a 19xx rule in asterisk and realized when I was trying to dial an area code 978 in MA that that was not a good idea. Is there a more defined rule for 900 space of non pay vs. pay codes? _1900NXX _NXX976 ___ --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] Dual Wan Router with Failover
Are you looking for load balancing or failover. Also is there a cheaper way of implementing load balancing than $845 appliance? Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd- Asterisk Sent: Tuesday, 14 November 2006 9:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dual Wan Router with Failover I've been looking for this as well.. I need to support up to 20 VOIP phones over Internet as the Asterisk server is off-site. We'll have multiple cable modems or DSL routers. I found this device which looks promising - does anyone have any experience with this? http://www.peplink.com/productsLoader.php?productName=balance Todd On Nov 13, 2006, at 8:49 PM, Dovid B wrote: Hi List, Does anyone know of a good dual wan router that can handle SIP well and can failover between connections if there is a SIP issue on one of the lines (meaning there still is a connection however there isnt enough bandwith or sip packets arent going thru etc.) ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel and limiting number off channels channels
Hi all, I have one interface zaptel TE110P with 10 channels enabled for my service provider. Alone the first channel is enable to receive call from this service provider and when a make call using other channel, like Zap/2, the channel is enable from the PBX Siemens EWSD version 10 to receive every calls. The support off my service provider, send to me the follow message that receive in their PABX: First DBAC e the last BPRN. somebody already had this problem and could help me? Tks, Rodrigo R Passos My configuration off zapata.conf is: [channels] language=br switchtype=euroisdn resetinterval=never signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes amaflags=billing immediate=no pridialplan=unknown callprogress=no callerid=asreceiveid txgain=5 callgroup=1 pickupgroup=1 context=EntradaPSTN group=1 channel=1-10 My configuration off zaptel.conf is: span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 loadzone=br defaultzone=br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Here's a question maybe someone can help me with: My extension looks like this: exten => 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3 When I try this extension, the following output appears in the CLI: Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored out with 0 I should mention that mpg123 is installed, however the server we are using for this project doesn't have an audio card. Is this a problem? Doesn't seem to be so far (everything else works great.) Cheers! Phil Jackson -- Phil Jackson CTO, Chesapeake Medical Imaging CEO/President, SecureRAD, LLC Tel: +1 443-716-0410 GSM: +1 202-841-0090 E-mail: [EMAIL PROTECTED] ___ --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] Dual Wan Router with Failover
We've never used it in a load balancing situation but it works great in a failover config. - Original Message - From: Dean Collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 14, 2006 10:41 AM Subject: RE: [asterisk-users] Dual Wan Router with Failover Sweet, now that is interesting http://hotbrick.com/produto.asp?tipo=2&codPro=22 anyone have any comments on the load balancing capability of these? ___ --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] Re: Is asterisk able to integrate with MS SQL
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield <[EMAIL PROTECTED] > wrote: In article <[EMAIL PROTECTED]>,Sharon Lim < [EMAIL PROTECTED]> wrote:>> Hi there,>> I am looking around, is there anyone did any integration asterisk talk to /> connect to MS SQL?Look for the package FreeTDS and install it. Then build Asterisk and it will include the TDS driver that can log CDRs to MS SQL.Alternatively, install ODBC drivers for MS SQL and then use Asterisk's ODBCfunctions.CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
