[asterisk-users] smartphone linked servers
We have 2 linked servers (IAX) and I would like our smartphone to be able to show the line status of an extension on the linked server. Meaning if an extension on the linked server is being used, I want the light corresponding to that extension to be lit up on the smartphone the same way it works if an extension on the same server is being used. Thank You Sim Zacks IT Manager Compulab 04-829-0145 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
On 17:54, Mon 08 Oct 07, D4rk F1ber wrote: So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. What did the trick for me is integrating it with MythTV. When the phone rings my tv pauses, and starts recording on the harddisk. Once the call is over my wife has 15 seconds to go back to her seat before the tv resumes. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
D4rk F1ber wrote: snip / One of the next projects for me personally is to get a SIP client for my Cingular/ATT 8525, it has wifi and hsdpa running Windows Mobile 6 and I am certain I have run across SIP clients before for these things. Be fun to play with and get working. So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. I run a small Open Source consulting/training company here in the Uk and am starting to build an * server so that myself and my business partner (who both work from our respective homes) are communicating properly. I have an analogue line coming into my home-office which connects to an x100p clone. Our plan is to be able to use that number for several of our business ventures (we have a couple of others between us :) and calls can be routed to our local handsets or voicemail or perhaps, to our mobiles or WiFi phones in the future... It's an interesting project, which serves two purposes for me. 1, We get an advanced, networked PBX system for a 2 man company :-) 2, We get to learn about using/deploying asterisk so we can advocate it in our business discussions. There's no better way to learn about something than by using it :-) My plan for the unit at my house, being a low power device, is to install something called Untangle (a fairly recently Open Sourced security platform), alongside Asterisk and Samba for a 24/7 home server and web filter/cache/firewall etc. (Possibly I'll add a UPnP backend if I have any grunt left in the machine). I'm blogging about it as I go if anyone is interested. Here's the first part of the story: http://www.theopensourcerer.com/2007/09/08/untangle-asterisk-pbx-and-file-server-all-in-one/ Cheers Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
On 10/9/07, D4rk F1ber [EMAIL PROTECTED] wrote: So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. I have a very simple setup for my Asterisk PBX at home. It comes with a Digium Dev Kit with 1 FXO and 1 FXS. It is also peered to SIPphone and FWD. All the extension numbers comes with a voicemail where the voicemail messages are sent to the individual e-mail accounts rather than storing them on the local hard disk drive of the server. Lastly, meetme is enabled so that if there will be at least 3 of us who are going to chat at the same time, at least we can do it easily. By the way, my Asterisk PBX server is also my wireless access point, web server, file server, music server, VPN server, database server, firewall and router. GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
Michiel van Baak wrote: snip / What did the trick for me is integrating it with MythTV. When the phone rings my tv pauses, and starts recording on the harddisk. Once the call is over my wife has 15 seconds to go back to her seat before the tv resumes. Way cool :-) -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
Alan Lord wrote: I run a small Open Source consulting/training company here in the Uk and am starting to build an * server so that myself and my business partner (who both work from our respective homes) are communicating properly. I have a couple of colleagues who also work from home - they're hooked into our office telephone system (Asterisk box) using SIP phones from their respective home offices. This way they are virtually in the office - external calls can be forwarded 'internally', and when they call customers, it looks as if they're calling from the office. It also means that our main lines carry all the calling costs, so no extra bills or expenses to deal with. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best config for 12 FXO system?
Maybe I was lucky, but a client of mine has a 24 FXO TDM2400 and works like a charm :) l. On Sun, 07 Oct 2007 03:06:52 +0200, C F [EMAIL PROTECTED] wrote: Because they tried competing with the channel bank market. But guess what, it has only one competitive edge, it's cheaper. But if you want something that works use a channel bank. Although I have no experience with Xorcoms USB based channel banks, I have a feeling they work better than Digiums TDM24xx cards. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
I understand you - it's better to settle down for a few hours with a book of the dead-tree kind. :) You could also try SIP Beyond VoIP - it's not just on SIP, but it gives you a broader usage/adoption scenario. l. On Mon, 08 Oct 2007 13:42:01 +0200, Justin Case [EMAIL PROTECTED] wrote: If I am behind the computer I end up just working. I need to get away and read the book. Only way I will really learn ;) On 10/7/07, Steve Totaro [EMAIL PROTECTED] wrote: Justin Case wrote: Hi List, I am trying to learn SIP in its entirety. I have so far found: http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 Anyone know of any other books that are worth reading ? Thanks. Justin The RFCs are online as well as anything else you could want to know. Are you just a book person? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice server
On Tue, 09 Oct 2007 01:05:41 +0200, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Asterisk can do all of that. Something along the lines of Thanks a lot for the help :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] which pci has the dell / hp
I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) Julian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which pci has the dell / hp
On 10/9/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) i have tested 3.3v but PCI V 3 if PCI2.X will not work. ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
D4rk F1ber wrote: So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. I do trunks/terminations so its easy for me to set all sorts of fun things up. Anyhow, here is a method for pitching it to your wife. If you have family dispersed throughout the United States, get yourself an 800 number and let Asterisk manage the way your family connects to each other at a cheap rate: E.g. Example: Mom 12125551000 (New York) Dan 13015551001 (DC) Tom 19085552001 (Jersey) Create a dialplan so your family can call your 800 number then re-route them to the family member of choice for example: Press 1 for Mom, Press 2 for Dan and so on... [transfer] exten = 1,1,Dial(SIP/[EMAIL PROTECTED]) ; Call Mom exten = 2,1,Dial(SIP/[EMAIL PROTECTED]) ; Call Dan exten = 3,1,Dial(SIP/[EMAIL PROTECTED]) ; Call Tom Since you stated something about a child, this would also help them in the unfortunate event of them either not having a cellular nor money. They can call you toll free... You can create a find me follow me context and have a context ring multiple numbers... Send telemarketers to telemarketer hell on transfer (http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture) There is a lot of nifty stuff you can do. If you're willing to get some ATA's dirt cheap and you have family abroad, you can save your entire family money. There are many things you could do with it on a personal level. J. Oquendo Excusatio non petita, accusatio manifesta http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xF684C42E sil . infiltrated @ net http://www.infiltrated.net smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DS3 Interface
If it hasn't already been done I am looking to put together a team to write drivers for this DS3 card to interface asterisk. http://www.imagestream.com/PCI_921-CDS.html The card itself seems reasonable and I believe we can make it work. As soon as I have positive feedback to begin the project I will put a server on the net with a card in it. Let's make this happen. Tim King CEO http://www.compnetwork.net/ CNS_LOGO_Beveled 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net/ http://www.compnetwork.net image001.png___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Atomic extensions reload
Hello everybody, is it possible that, when Asterisk is executing extensions reload, if I issue another extensions reload I can mess up the dialplan? If so, I think that the correct behaviour should be using a lock for the dialplan and letting the second extensions reload wait for the first to finish the execution. Should I file a bug? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? On 10/9/07, Tim King [EMAIL PROTECTED] wrote: If it hasn't already been done I am looking to put together a team to write drivers for this DS3 card to interface asterisk. http://www.imagestream.com/PCI_921-CDS.html The card itself seems reasonable and I believe we can make it work. As soon as I have positive feedback to begin the project I will put a server on the net with a card in it. Let's make this happen. *Tim King* *CEO* [image: CNS_LOGO_Beveled] http://www.compnetwork.net/ 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image001.png___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] advice on sip
Hi if i want to use sip client to connect to my asterisk pbx do i need to run a sip server ? If so can you point me in the direction of a good howto for asterisk and sip ... Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server
I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good Book to learn SIP
I think that reading an introductory book AND the rfc is the best choice to learn sip. The rfc is very well written and is a more complete reference. Wiresharking sip conversations could help you too. On 10/9/07, Lenz [EMAIL PROTECTED] wrote: I understand you - it's better to settle down for a few hours with a book of the dead-tree kind. :) You could also try SIP Beyond VoIP - it's not just on SIP, but it gives you a broader usage/adoption scenario. l. On Mon, 08 Oct 2007 13:42:01 +0200, Justin Case [EMAIL PROTECTED] wrote: If I am behind the computer I end up just working. I need to get away and read the book. Only way I will really learn ;) On 10/7/07, Steve Totaro [EMAIL PROTECTED] wrote: Justin Case wrote: Hi List, I am trying to learn SIP in its entirety. I have so far found: http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403 Anyone know of any other books that are worth reading ? Thanks. Justin The RFCs are online as well as anything else you could want to know. Are you just a book person? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd router behavior when using 'w' in SendDTMF
Hey, This is weird, I wonder if anyone has an explanation? If I call a SIP server and inject DTMF with a wait in it, my router will then lock up causing asterisk to lose Internet connectivity obviously, but also making it very hard to see what happens. It appears that if there are no 'w' in the DTMF string, it doesn't lock up. Anyone have any guesses on this? I called a local extension, and the tones sound perfectly normal and are delayed just the right amount and played properly to the channel. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? I can imagine it be used as a TDM-SIP gateway but if I needed such a box I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or look at FreeSWITCH which by design seems more suitable for these kind of high performance applications. There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Why wouldn't today's powerful quadcore servers with Gigabit Ethernet interfaces not be able to handle less than 100Mbit/s synchronous traffic? Please enlighten me as I am no expert here. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
A few SGI boxen with Numalink could probably handle it just fine. Thanks, Steve Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? On 10/9/07, *Tim King* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If it hasn't already been done I am looking to put together a team to write drivers for this DS3 card to interface asterisk. http://www.imagestream.com/PCI_921-CDS.html http://www.imagestream.com/PCI_921-CDS.html The card itself seems reasonable and I believe we can make it work. As soon as I have positive feedback to begin the project I will put a server on the net with a card in it. Let's make this happen. *Tim King* *CEO* CNS_LOGO_Beveled http://www.compnetwork.net/ 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net http://www.compnetwork.net/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
It's not the Ethernet interface that would be the issue. The zaptel framework wouldn't be able to handle it with the way it uses interrupts. On 10/9/07, Patrick [EMAIL PROTECTED] wrote: On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? I can imagine it be used as a TDM-SIP gateway but if I needed such a box I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or look at FreeSWITCH which by design seems more suitable for these kind of high performance applications. There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Why wouldn't today's powerful quadcore servers with Gigabit Ethernet interfaces not be able to handle less than 100Mbit/s synchronous traffic? Please enlighten me as I am no expert here. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE : Re: [asterisk-dev] oh323.conf, extentions.conf
You don't need to define a gatekeeper, it's optional.
It's not official documentation and not prove that, although I think you
could believe in it.
http://www.voip-info.org/wiki/view/Asterisk+oh323+channels
Regards
On 10/9/07, brahem mehdi [EMAIL PROTECTED] wrote:
thanks Machado,
but i have one question
this line i have to define the gatekeeper
gatekeeper=A.B.C.D ; GnuGK
so Asterisk is can not a gatekeeper with H323 ??
have you an official document to prove that??
thanks again
*Caciano Machado [EMAIL PROTECTED]* a écrit :
Send these questions to Asterisk-Users mailing list.
h323.conf
##
;
; Configuration file of OpenH323 channel driver
;
[general]
listenAddress=W.X.Y.Z ; local ip
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=100
inboundMax=100
simultaneousMax=100
wrapLibTraceLevel=0
libTraceLevel=0
libTraceFile=stdout
gatekeeper=A.B.C.D ; GnuGK
gatekeeperTTL=600
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
; INBAND -
;
userInputMode=TONE
amaFlags=default
accountCode=H323
language=en
musiconhold=default
;context=from-h323-filter
context=from-h323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
alias=GW-PABX
gwprefix=3308
gwprefix=0514
gwprefix=0513
gwprefix=0512
;-
; Specify and configure CODEC related
; options
;-
[codecs]
codec=G711U
frames=20
codec=G711A
frames=20
#
extensions.conf
#
...
exten = _0XX1XXX,1,Dial(OH323/${EXTEN},20,rtT)
...
#
On 10/5/07, brahem mehdi wrote:
hi all,
can any one give me a correct configuration for the 2 files oh323.confet
extensions.conf .
i work on VOIP ( asterisk et oh323) but i can't make it work.
thanks
---
Brahem mehdi
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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Re: [asterisk-users] get egress SIP call Id
Hi Philipp, Thank you for your response to my question. I am working on a project which uses Asterisk as the voice engine. I need to get the ingress and egress sip call id for a call to write call CDR. (Asterisk CDR does not meet our customer requirments). If there is no any easy way to get it I might need to create a seperate process/thread to query manager interface as you mentioned. Thanks you, Ray Ray Chen wrote: Hi, Does anybody know how to get the SIP call ID of a Dial command? There's no easy way to do it. What's your intention? There are several events on the manager interface. Regards, Philipp Kempgen -- -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] advice on sip
On 10/9/07, Gregory Machin wrote: Hi if i want to use sip client to connect to my asterisk pbx do i need to run a sip server ? If so can you point me in the direction of a good howto for asterisk and sip ... install any sip client on your workstation computer and point it to your asterisk box IP address, you should hear the welcome to asterisk message. http://www.asterisk.org/support/get-started I suggest visting the library or purchasing : Asterisk : The Future of Telephony, 2nd edition ISBN 10: 0-596-51048-9 ISBN 13: 9780596510480 and reading up a bit. http://www.oreilly.com/catalog/9780596510480/index.html Specifically /etc/asterisk/sip.conf defines the aspects of SIP channles on your asterisk box along with some port parameters from rtp.conf -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I did not look at the specs of the card but if it has inboard DSPs, it may work just fine in a high end box. Thanks, Steve Matt wrote: It's not the Ethernet interface that would be the issue. The zaptel framework wouldn't be able to handle it with the way it uses interrupts. On 10/9/07, * Patrick* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? I can imagine it be used as a TDM-SIP gateway but if I needed such a box I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or look at FreeSWITCH which by design seems more suitable for these kind of high performance applications. There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Why wouldn't today's powerful quadcore servers with Gigabit Ethernet interfaces not be able to handle less than 100Mbit/s synchronous traffic? Please enlighten me as I am no expert here. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T-Mobile and WiFi Voip
I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] get egress SIP call Id
You can capture the sipcallid from the manager output. The cool part is that the sipcallid is the same on both sides of a call. So, AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as AsteriskA for that call. It is really easy to capture it from the manager. Thanks, Steve Ray Chen wrote: Hi Philipp, Thank you for your response to my question. I am working on a project which uses Asterisk as the voice engine. I need to get the ingress and egress sip call id for a call to write call CDR. (Asterisk CDR does not meet our customer requirments). If there is no any easy way to get it I might need to create a seperate process/thread to query manager interface as you mentioned. Thanks you, Ray Ray Chen wrote: Hi, Does anybody know how to get the SIP call ID of a Dial command? There's no easy way to do it. What's your intention? There are several events on the manager interface. Regards, Philipp Kempgen -- T ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T-Mobile and WiFi Voip
its IMS /b On Oct 9, 2007, at 10:39 AM, Andres wrote: I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
http://www.imagestream.com/PCI_921-CDS.html This card can do it. I have spoke with them about it and its very capable of doing what is needed for a DS3 in a standard linux box. /b On Oct 9, 2007, at 10:42 AM, Andrew Kohlsmith wrote: On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Patrick wrote: On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? I can imagine it be used as a TDM-SIP gateway but if I needed such a box I'd rather go for a Lucent MaxTNT, Lucent APX8000 or a Cisco 5xxx or look at FreeSWITCH which by design seems more suitable for these kind of high performance applications. There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Why wouldn't today's powerful quadcore servers with Gigabit Ethernet interfaces not be able to handle less than 100Mbit/s synchronous traffic? Please enlighten me as I am no expert here. Regards, Patrick Perhaps it could also be used as a pure TDM switch with no VoIP calls involved? Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server
Steve Totaro wrote: I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro I think I may have figured out my own issue. Since I am creating multiple SIP peers on two boxes that point to each other, I need to define separate ports for each one. Anyone know if that is the case? Makes sense to me but I cannot try it on the live server and my dev boxes are all doing other things. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Andrew Kohlsmith wrote: On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. A single core HPDL320 with core solo 3ghz would hit 60%-70% (which I felt comfortable with but did not want to go over) CPU in top terminating four PRIs with all channels in use. The box did nothing but take the voice PRI and put on the LAN as ulaw, so no transcoding. You would need a very beefy server, thats for sure. I the board had no onboard processing or DSPs, you would probably need something 7x more powerful than I listed above. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outside queue members not ringing.
On 10/8/07, Alex Balashov [EMAIL PROTECTED] wrote: Greetings, I have a very basic equal-weight ring-all queue set up in queues.conf: [sales-queue] ;music = default strategy = ringall periodic-announce-frequency = 20 announce-holdtime = no timeout = 15 maxlen = 0 member = SIP/[EMAIL PROTECTED],1 member = SIP/[EMAIL PROTECTED],1 Are you using masks to the queue extensions in queues.conf or it's just a generic example? If it's a mask try to dump the sip invite To: field and check if it's the correct destination. I don't know if asterisk support masks there. member = SIP/dude,1 member = SIP/homie,1 member = SIP/fellow,1 But for some reason, the calls to the outside SIP parties never seem to go out, if they ever did before. I've been running 1.4.x for a long time. A packet capture reveals that no SIP INVITE goes to the junction_networks peer at all, even though it is available and qualified as reachable. Anyone know what gives? Cheers, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T-Mobile and WiFi Voip
Yep all the carriers are looking to offer 'voip' services sooner rather than later. Basically it uses the wifi point to access the mobile switching network. Cool part is you will soon be answering your Verizon home phone on your cell when you are 'within range' or your home network. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, 9 October 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T-Mobile and WiFi Voip I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
This looks very promising. All eggs in one basket, but promising... Any idea on price? The PCI 921-CDS utilizes the Mindspeed CX28500 chipset to provide support for the card's host PCI bus interface, which can burst data at speeds up to 780 Mbps, or 390 Mbps full duplex. The CX28500 also provides the card's DMA controller, HDLC controllers, and management interface. The CX28500 is connected to the PMC-Sierra PM8315, which provides the card's T1 framers and M13 multiplexer. The PM8315 is also connected to the Exar XRT73L00 T3 LIU, which supports the physical DS3 line interface to the card. Thanks, Steve Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html This card can do it. I have spoke with them about it and its very capable of doing what is needed for a DS3 in a standard linux box. /b On Oct 9, 2007, at 10:42 AM, Andrew Kohlsmith wrote: On Tuesday 09 October 2007 10:14:23 Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? Whatever gave you the notion that a modern PC can't handle 672 simultaneous calls? -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On 10/9/07, Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html [...] off-topic : I saw Imagestream at the Ohio Linuxfest a weekend ago. Also picked up a few literature bags by Digium :-) -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T-Mobile and WiFi Voip
Will this work backwards? When I'm at home instead of my cell ringing have the home phone ring? Why would anyone give up revenue from minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, October 09, 2007 12:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T-Mobile and WiFi Voip Yep all the carriers are looking to offer 'voip' services sooner rather than later. Basically it uses the wifi point to access the mobile switching network. Cool part is you will soon be answering your Verizon home phone on your cell when you are 'within range' or your home network. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, 9 October 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T-Mobile and WiFi Voip I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server
Steve Totaro wrote: Steve Totaro wrote: I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro I think I may have figured out my own issue. Since I am creating multiple SIP peers on two boxes that point to each other, I need to define separate ports for each one. Anyone know if that is the case? Makes sense to me but I cannot try it on the live server and my dev boxes are all doing other things. no. It might be the case if you had multiple SIP clients behind the same NAT router connection to a non-local Asterisk box. The userid and password that is sent with the call should make it hit the correct sip.conf entry. Perhaps you are doing something silly in your sip.conf configs. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error: 603 declined
I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication 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 [user1] type=friend username=user1 secret=xxx host=dynamic context=work [user2] type=friend username=user2 secret=xxx host=dynamic context=work extensions.conf: [work] exten = ,1,Dial(SIP,user1) exten = 1112,1,Dial(SIP,user2) When we use Twinkle as our SIP client, user1 calls user2 dialing and user2 calls user1 dialing 1112, we get this error: Line 1 Call failed - 603 declined.so I can make a call. In Asterisk I debug the channel and I get this log: voip*CLI debug channel 1 No such channel 1 Debugging on new channels is enabled -- Executing Dial(SIP/user1-08148450, SIP|user2) in new stack Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1) == Spawn extension (sintys, 1112, 1) exited non-zero on 'SIP/user1-08148450' Oct 9 12:52:41 WARNING[3453]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81508e8', 10 retries! What is the problem ??? Any help please ??? Thanks a lot Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I have started the open source project to get this going. I am working directly with the manufacture to form agreements and gain access to the hardware and source code for their drivers. The list price for the card is $4,995.00 USD. I will keep everyone posted and will have site for development and forums up soon. Thanks for the support Tim King CEO 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, October 09, 2007 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DS3 Interface On 10/9/07, Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html [...] off-topic : I saw Imagestream at the Ohio Linuxfest a weekend ago. Also picked up a few literature bags by Digium :-) -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server
Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro I think I may have figured out my own issue. Since I am creating multiple SIP peers on two boxes that point to each other, I need to define separate ports for each one. Anyone know if that is the case? Makes sense to me but I cannot try it on the live server and my dev boxes are all doing other things. no. It might be the case if you had multiple SIP clients behind the same NAT router connection to a non-local Asterisk box. The userid and password that is sent with the call should make it hit the correct sip.conf entry. Perhaps you are doing something silly in your sip.conf configs. Perhaps I am, let's hope so. This was my latest attempt to get it to work. The other server looks identical except the host IP. [general] ;bindport=5060 bindaddr=0.0.0.0 [default] [span1] type=friend host=192.168.6.2 username=span1 secret= context=to-span1 auth=rsa inkeys=span1-2-fast1 outkey=fast1-2-span1 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very [span2] type=friend host=192.168.6.2 username=friend secret=x context=to-span2 auth=rsa inkeys=span2-2-fast1 outkey=fast1-2-span2 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very [span3] type=friend host=192.168.6.2 username=span3 secret=x context=to-span3 auth=rsa inkeys=span3-2-fast1 outkey=fast1-2-span3 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very [span4] type=friend host=192.168.6.2 username=span4 secret=x context=to-span4 auth=rsa inkeys=span4-2-fast1 outkey=fast1-2-span4 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error: 603 declined
This line gives you the clue: Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1 Your dialplan should have Dial(SIP/user1) rather than Dial (SIP,user1) / instead of , Give that a try. -- Aubrey Wells Senior Engineer Shelton | Johns Technology Group www.sheltonjohns.com On Oct 9, 2007, at 12:05 PM, Alejandro Cabrera Obed wrote: Oct 9 12:52:41 WARNING[3478]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] EM Wink and T4xxP losing ability to dial
Still having no luck with this scenario. Has anyone else experienced problems with em wink lines? I'm thinking that there could be problems with the timing settings in zapata.conf, but documentation is pretty light. How could the telco not be receiving enough digits when it works for 500 calls...? Anyone have any similar experiences/ possible solutions? Whit On 10/2/07, Whit Thiele [EMAIL PROTECTED] wrote: Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 EM wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using ZapBarge to listen to the channels themselves you can hear the dtmf digits being dialed, but then either there is dialtone (again), or a fast busy signal. I've also noticed(via ZapBarge) that before the T1 seizure that the call launched will immediately go to an automated message claiming that not enough digits have been dialed. The rest of the T1's become useless right after it. Once the T1's get into this state, even trying to launch a call directly through the dialplan experiences the same behavior. Its like the T1 has locked up. The telco provider (quest) claims they aren't even seeing the digits placed on the circuit. They say they see the channel being grabbed, but no digits appearing. Verbose and debug show nothing out of the ordinary. Could this be an issue with the Digium card? Both T1's experience the same behavior at the same time. I've tried some different settings such as callprogress=yes and busydetect=no but nothing has helped. Only restarting asterisk seems to allow another 600 calls to be processed. I'm using the latest asterisk release version 1.4.11 and zaptel 1.4.5.1 Regards, Whit ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
On Tue, 2007-10-09 at 09:55 +0200, Michiel van Baak wrote: On 17:54, Mon 08 Oct 07, D4rk F1ber wrote: I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. What did the trick for me is integrating it with MythTV. I have tried to get MythPhone to work without much success so far; maybe it's time to give it another try, because: When the phone rings my tv pauses, and starts recording on the harddisk. Once the call is over my wife has 15 seconds to go back to her seat before the tv resumes. ...this is exactly what I want. I originally started with * because when I upgraded my home server to new hardware, I could no longer use my ISA modem, and I experienced literally months of frustration trying to find a PCI modem that wasn't a Lose Modem (erroneously called Winmodem :-) or one that had a driver that would actually work with the vgetty+sendfax-based answering machine I had. I read an article in Linux Journal about * and decided to see what I could do with it. I also had a lot of time at home recovering from surgery and this gave me something to do. Now, using * and a $300 Digium card as an answering machine is massive overkill, so I assumed there would be other things I could do with it. I was right. One thing it does permit is the use of VoIP phones in the house, so I could install phones in places where there was network wiring but no phone wiring. Also soft phones on my laptop and desktop. It permits using the house phones as an intercom. It allows my wife and I to have separate voice mail boxes, plus one for a political organization we are involved with. We can have our voice messages e-mailed (very handy when we are on trips). I can program it to prevent my wife from lapsing into old habits and making long distance calls on the house line (and activating a monthly fee) when we have prepaid long distance on our cell phones. We can record calls. We can access the answering machine from any phone in the house. It will automatically route incoming faxes to the machine with the fax modem, so there is no need for a separate line or always having to make special arrangements for faxes. At some future time, having the ability to receive calls over the Internet via services such as FWD will come in handy (when that becomes popular enough to be more than a fun toy for geeks). There are enough advantages that, even though the phones may be a bit more difficult to use now (you can't just pick up another extension, you have to initiate a conference call or transfer the call), she has warmed up to * (and MythTV) because of the additional features it offers over an old-fashioned answering machine. --Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I'm already doing that. /b On Oct 9, 2007, at 11:31 AM, Tim King wrote: I have started the open source project to get this going. I am working directly with the manufacture to form agreements and gain access to the hardware and source code for their drivers. The list price for the card is $4,995.00 USD. I will keep everyone posted and will have site for development and forums up soon. Thanks for the support Tim King CEO 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, October 09, 2007 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DS3 Interface On 10/9/07, Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html [...] off-topic : I saw Imagestream at the Ohio Linuxfest a weekend ago. Also picked up a few literature bags by Digium :-) -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] EM Wink and T4xxP losing ability to dial
A critical lesson I learned was not to rely to heavily on the AMI, especially when there are other ways of doing the same thing that are just as simple. I suggest, rather than using AMI originate, mv or ftp .call files. Thanks, Steve Whit Thiele wrote: Still having no luck with this scenario. Has anyone else experienced problems with em wink lines? I'm thinking that there could be problems with the timing settings in zapata.conf, but documentation is pretty light. How could the telco not be receiving enough digits when it works for 500 calls...? Anyone have any similar experiences/ possible solutions? Whit On 10/2/07, *Whit Thiele* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 EM wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using ZapBarge to listen to the channels themselves you can hear the dtmf digits being dialed, but then either there is dialtone (again), or a fast busy signal. I've also noticed(via ZapBarge) that before the T1 seizure that the call launched will immediately go to an automated message claiming that not enough digits have been dialed. The rest of the T1's become useless right after it. Once the T1's get into this state, even trying to launch a call directly through the dialplan experiences the same behavior. Its like the T1 has locked up. The telco provider (quest) claims they aren't even seeing the digits placed on the circuit. They say they see the channel being grabbed, but no digits appearing. Verbose and debug show nothing out of the ordinary. Could this be an issue with the Digium card? Both T1's experience the same behavior at the same time. I've tried some different settings such as callprogress=yes and busydetect=no but nothing has helped. Only restarting asterisk seems to allow another 600 calls to be processed. I'm using the latest asterisk release version 1.4.11 and zaptel 1.4.5.1 http://1.4.5.1 Regards, Whit ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve Brian West wrote: I'm already doing that. /b On Oct 9, 2007, at 11:31 AM, Tim King wrote: I have started the open source project to get this going. I am working directly with the manufacture to form agreements and gain access to the hardware and source code for their drivers. The list price for the card is $4,995.00 USD. I will keep everyone posted and will have site for development and forums up soon. Thanks for the support Tim King CEO 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Baji Panchumarti Sent: Tuesday, October 09, 2007 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DS3 Interface On 10/9/07, Brian West wrote: http://www.imagestream.com/PCI_921-CDS.html [...] off-topic : I saw Imagestream at the Ohio Linuxfest a weekend ago. Also picked up a few literature bags by Digium :-) -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] EM Wink and T4xxP losing ability to dial
Just to be clear, I would eliminate the AMI as the culprit first. I have seen extensive use of the AMI cause all kinds of flaky behavior. Zaptel, timing, or EM wink may be working perfectly but the AMI is borking everything up, thats my thought anyways. Thanks, Steve Totaro Steve Totaro wrote: A critical lesson I learned was not to rely to heavily on the AMI, especially when there are other ways of doing the same thing that are just as simple. I suggest, rather than using AMI originate, mv or ftp .call files. Thanks, Steve Whit Thiele wrote: Still having no luck with this scenario. Has anyone else experienced problems with em wink lines? I'm thinking that there could be problems with the timing settings in zapata.conf, but documentation is pretty light. How could the telco not be receiving enough digits when it works for 500 calls...? Anyone have any similar experiences/ possible solutions? Whit On 10/2/07, *Whit Thiele* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 EM wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using ZapBarge to listen to the channels themselves you can hear the dtmf digits being dialed, but then either there is dialtone (again), or a fast busy signal. I've also noticed(via ZapBarge) that before the T1 seizure that the call launched will immediately go to an automated message claiming that not enough digits have been dialed. The rest of the T1's become useless right after it. Once the T1's get into this state, even trying to launch a call directly through the dialplan experiences the same behavior. Its like the T1 has locked up. The telco provider (quest) claims they aren't even seeing the digits placed on the circuit. They say they see the channel being grabbed, but no digits appearing. Verbose and debug show nothing out of the ordinary. Could this be an issue with the Digium card? Both T1's experience the same behavior at the same time. I've tried some different settings such as callprogress=yes and busydetect=no but nothing has helped. Only restarting asterisk seems to allow another 600 calls to be processed. I'm using the latest asterisk release version 1.4.11 and zaptel 1.4.5.1 http://1.4.5.1 Regards, Whit ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T-Mobile and WiFi Voip
Technically anything is possible - a few years ago I was working with Siemens to implement something called Openscape which never took off in the USA but basically was a web based application which allowed company users to redirect their office phone numbers from the web to their mobile or home numbers etc (also had some UM features as well). Point being is technically anything is possible it's just a commercial decision on how things are offered. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Aarons (US) Sent: Tuesday, 9 October 2007 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [asterisk-users] T-Mobile and WiFi Voip Will this work backwards? When I'm at home instead of my cell ringing have the home phone ring? Why would anyone give up revenue from minutes? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, October 09, 2007 12:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T-Mobile and WiFi Voip Yep all the carriers are looking to offer 'voip' services sooner rather than later. Basically it uses the wifi point to access the mobile switching network. Cool part is you will soon be answering your Verizon home phone on your cell when you are 'within range' or your home network. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, 9 October 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T-Mobile and WiFi Voip I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weatherproof Hard Phone
Hello Don, thanks for the helpful pointers, i'll push my quotes on these and hopefully they will be accepeted. The only drawback on this is the fact that i would have to use an ATA to complete the loop. This will rais the unit cost of the deployment. I was thinking of usin SOEKRIS installed with asterisk, though i doubt this would be a cheaper solution On 10/8/07, Don Kelly [EMAIL PROTECTED] wrote: For discussing financial transactions, I think a handset would be required. For example, search MIS6I on this page: http://www.sandman.com/autodial.html Use it with your favorite ATA. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Monday, October 08, 2007 8:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Weatherproof Hard Phone Use a linksys speakerphone behind a metal late and a Push to Connect button wired as the hook switch - bat phone connection. Mount the mic and speaker on holes in the plate and the guts glued to the plate. Simple and cheap and they have to buy from you. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Don't take it personally. I have been on this list about as long as you. BKW (Next!) Ego can be good but let's not become egomaniacs shall we? I am not working on it from any angle, and would probably never *entertain* using such a device. I prefer tried and true DS3 MUXs such as the Adtran MX2800 and multiple TDM-SIP servers. So much built in failover in this approach short of the entire DS3 going down. Anyways, good luck with all 672 eggs in one basket, just don't drop it! It would suck to drop 600 plus calls all at once. Thanks, Steve Brian West wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
BTW, this is the wrong list if it not for Asterisk. It has absolutely nothing to do with Asterisk. Please post to the appropriate FreeSwitch list. Thanks again, Steve Totaro Brian West wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Well we are plugging it in the OpenZAP abstraction layer we have already started on. This is usable by Asterisk also so asterisk would benefit from it. http://fisheye.freeswitch.org/browse/OpenZAP /b On Oct 9, 2007, at 12:31 PM, Steve Totaro wrote: BTW, this is the wrong list if it not for Asterisk. It has absolutely nothing to do with Asterisk. Please post to the appropriate FreeSwitch list. Thanks again, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind Multi-NAT question
Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
And what was the purpose of this? /b On Oct 9, 2007, at 1:32 PM, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Yes, I knew who I was talking to and now I know a little more about you Matt, that was totally uncalled for. Thanks, Steve Totaro Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, *Brian West* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Matt, I talk very openly about this issue. It was very rude of you to bring this up. This plea was total bullshit. If you want to know the whole story feel free to call me and talk about it. 918-424-9378... anyone can call me and ask me questions about it. The plea was a deal worked out between the DOJ and my attorney which was good because I signed my plea on Sept. 4th 2001. If you try to fight the DOJ you will not win. That plea was the only way to settle the issue without a trial. All I did was click edit in frontpage and alert them of anonymous publishing priv. were on their servers and they called the FBI and three days later our office was raided. This I consider mudslinging by you and wasn't very gentle man like. /b On Oct 9, 2007, at 1:32 PM, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Multi-NAT question
I have tried it with the best result of one way audio after spending a few days doing everything imaginable. This is the only scenario where I suggest using IAX. Thanks, Steve Totaro WipeOut wrote: Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On 10/9/07, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm Hey, I am not sure what your point is, are you trying to shame West on this list with your post ? He is a contributor to the asterisk movement, which is the purpose of these lists. This was uncalled for. -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Perhaps it was uncalled for. However, if I were to consider using FreeSwitch I would want to know who was/is behind it. On 10/9/07, Brian West [EMAIL PROTECTED] wrote: And what was the purpose of this? /b On Oct 9, 2007, at 1:32 PM, Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On 10/9/07, Matt [EMAIL PROTECTED] wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm Fascinating. Not really. Anyway, how is this related to Asterisk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging in Asterisk
Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. /b On Oct 9, 2007, at 2:12 PM, Matt wrote: Perhaps it was uncalled for. However, if I were to consider using FreeSwitch I would want to know who was/is behind it. On 10/9/07, Brian West [EMAIL PROTECTED] wrote: And what was the purpose of this? /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Well hopefully people can read between the lines.. I have talked about this issue in public many times and don't try to hide it but the plea isn't how it went down. /b On Oct 9, 2007, at 1:50 PM, Steve Totaro wrote: Yes, I knew who I was talking to and now I know a little more about you Matt, that was totally uncalled for. Thanks, Steve Totaro Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, *Brian West* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api- digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click to Talk Web Applications with Asterisk
Hi, I would like to develop a click to talk app to interface with asterisk, anyone know about some SDK/frameworks to implement this. Regards. Ricardo Meléndez Rosales ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone using SIP trunks from Time Warner Telecom?
On 10/8/07, Forrest Beck [EMAIL PROTECTED] wrote: I was told that Asterisk was supported when we looked at the service. Hey Forrest - thanks for the information. Might you be able to send along the contact information for the TW rep who told you that asterisk was supported? I've been in conversation with our Sales rep today, and he's quite adamant that they currently only support Cisco Call Manager and CCM Express. I believe they're using CCM to provice the SIP trunks - if this is indeed the case, I don't see interoperability with asterisk as a problem. Thanks -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Nick Couchman wrote: Our office does not have a PA system, and in our current phone system we have a certain extension that we dial that pages over the speaker of all the phones in the office. Does Asterisk support this feature? If so, could someone tell me the best way to set this up in AsteriskNOW? Thanks, Nick I could not tell you in asterisknow but I use this feature with Polycom phones on all of my installs. It is very well documented in voip-info.org Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
PLEASE, take the old jiaxclient code and bring it back to life! It had so much potential. Thanks, Steve Totaro Ricardo Melendez wrote: Hi, I would like to develop a “click to talk” app to interface with asterisk, anyone know about some SDK/frameworks to implement this. Regards. *Ricardo Meléndez Rosales* ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When does the future arrive?
Hi all, Probably this is the wrong place to ask, but is there an estimated time of arrival of the future? i.e. TFOT--next generation dealing with * -1.4 I attended a workshop some time ago, and the book was part of the package HtH, Hans ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Thomson ST2030 firmware upgrade
Hello, I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42 firmware to the latest version (1.56) through tftp. The phone loads the .inf file, then the correct firmware file (as stated in the ST2030S.inf), then it reboots and loops doing these same things again and again. The firmware version on the phone stays at 1.42. Is there a special intermediate firmware version to use before going to the latest? Something special to include in the .inf file? I looked everwhere on the Net (including voip-info). Thanks, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Multi-NAT question
Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut: Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? My experience with SIP, Asterisk and more than one NAT in the path is not a good one. For example, several of my SIP hardphones refused to work behind a dual-NAT Phone (10.10.0.201) - NAT - internal net (192.168.174.0/24) - NAT - Internet - Asterisk where everything else worked as usual. Admittedly multiple NATs are not necessarily a good idea to have, but that was a customer's network, not mine ;-) Also quite regular setups like Phone - NAT - Internet - NAT - Asterisk and 2 Phones - NAT - Internet - Asterisk without NAT (One of those phones calling the other). might work - or just be a source of trouble. This also seems to depend on the cooperation of the NAT device; some work better than others. IAX seems to handle NAT issues much better, in my experience, but I did never have an IAX hardphone. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does the future arrive?
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote: Hi all, Probably this is the wrong place to ask, but is there an estimated time of arrival of the future? i.e. TFOT--next generation dealing with * -1.4 I attended a workshop some time ago, and the book was part of the package The Future, my friend, is here. http://downloads.oreilly.com/books/9780596510480.pdf Enjoy! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez: Hi, I would like to develop a “click to talk” app to interface with asterisk, anyone know about some SDK/frameworks to implement this. I have not ever used such an application, but there are several solutions commercially available. If your intention is getting a solution, you might consider spending money. If your intention is learning, the better - but sorry, I cannot give adequate pointers there. I remember there were open source puzzles parts that could be mended to something like a web click-to-call app, might be the term jiaxclient relates to that. Do not count to much of that, my brain is getting old. I do not want to advertise a specific solution, but you could search the mailing list archives - click to call might be a subject worth reading. You could also look for something like IAX Client JAVA. I bet there is also some information to be found on voip-info.org. I think at least one vendor offers free trial versions so you could at least test wether the concept is viable, and then decide to either spend money or time on the project. I hope you did not trigger one of those Hey, I have a solution for you, hey, this is a non-commercial-list, go die flamewar - we had enough of those ;-) Best regards, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tue, 9 Oct 2007, Brian West wrote: I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. The reference to FreeSwitch was uncalled for. Posting a link to a fact is not passing judgement. (It may be unrelated to Asterisk, but it was of interest.) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Multi-NAT question
Anselm Martin Hoffmeister wrote: Am Dienstag, den 09.10.2007, 19:50 +0100 schrieb WipeOut: Hi, Ok.. I know dual NAT is a problem for SIP.. ie. UA - NAT - Internet - NAT - Asterisk What about Multi-NAT where a dedicated public IP is mapped to the private IP of the asterisk box.. ie UA - NAT - Internet - Multi-NAT - Asterisk http://www.draytek.co.uk/support/kb_vigor_multinat.html Anyone tried it? My experience with SIP, Asterisk and more than one NAT in the path is not a good one. For example, several of my SIP hardphones refused to work behind a dual-NAT Phone (10.10.0.201) - NAT - internal net (192.168.174.0/24) - NAT - Internet - Asterisk where everything else worked as usual. Admittedly multiple NATs are not necessarily a good idea to have, but that was a customer's network, not mine ;-) Also quite regular setups like Phone - NAT - Internet - NAT - Asterisk and 2 Phones - NAT - Internet - Asterisk without NAT (One of those phones calling the other). might work - or just be a source of trouble. This also seems to depend on the cooperation of the NAT device; some work better than others. IAX seems to handle NAT issues much better, in my experience, but I did never have an IAX hardphone. BR Anselm For a small investment of time and money, you can setup OpenVPN and have your own network with no NAT issues whatsoever. That would be my first choice over IAX. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote: Before you put any work into this... ask yourself... what exactly are you hoping to accomplish? There is no way one system can handle a DS3s worth of traffic... therefore, what good would this do? I presume you can compare it with an ETSI C3 (34Mb instead of 45Mb) Sometime offices are interconnected with them, and only a part of them (a quarter) is used for telephony. In such cases, you would handle 120 lines. And would latest systems not be able to cope with 480 lines? Those beasts like hp580-G5 are probably heavy enough (four quad cores, 128GB mem) Only fact against such config, is that it would be a major SPOF. HW ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Wow. It shows that there is a lot of ignorance in the DOJ. They should have thanked BW, not charged him. Thanks for blowing this way off track Matt. Tom At 01:32 PM 10/9/2007, you wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htmhttp://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, Brian West mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Multi-NAT question
On 16:32, Tue 09 Oct 07, Steve Totaro wrote: For a small investment of time and money, you can setup OpenVPN and have your own network with no NAT issues whatsoever. That would be my first choice over IAX. Or wait till the ipv6 branch is ready for production. NO MORE NAT ! YAY! -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound call voip providers
Rafael: Thanks for your reply. I browsed http://www.fonetglobal.com but it seems to have local numering only in America. We need this service but in Europe. Do you have this service in Europe? The thing that we need is pretty simple. When the user calls a normal PSTN phone# from his normal PSTN telephone the provider stablishes a SIP session over IP to our asterisk box. Regards On Monday 08 October 2007 23:08, Rafael Canchola wrote: http://www.fonetglobal.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help With Error
This is the first time that I am seeing this error. Can anyone help me with its meaning ? pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! Thanks. Dovid___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does the future arrive?
On Tue, 2007-10-09 at 15:29 -0500, Erik Anderson wrote: On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote: Hi all, Probably this is the wrong place to ask, but is there an estimated time of arrival of the future? i.e. TFOT--next generation dealing with * -1.4 I attended a workshop some time ago, and the book was part of the package The Future, my friend, is here. http://downloads.oreilly.com/books/9780596510480.pdf Enjoy! Tnx a lot! Will certainly do... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
Google for mexuar. Zoa Anselm Martin Hoffmeister wrote: Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez: Hi, I would like to develop a “click to talk” app to interface with asterisk, anyone know about some SDK/frameworks to implement this. I have not ever used such an application, but there are several solutions commercially available. If your intention is getting a solution, you might consider spending money. If your intention is learning, the better - but sorry, I cannot give adequate pointers there. I remember there were open source puzzles parts that could be mended to something like a web click-to-call app, might be the term jiaxclient relates to that. Do not count to much of that, my brain is getting old. I do not want to advertise a specific solution, but you could search the mailing list archives - click to call might be a subject worth reading. You could also look for something like IAX Client JAVA. I bet there is also some information to be found on voip-info.org. I think at least one vendor offers free trial versions so you could at least test wether the concept is viable, and then decide to either spend money or time on the project. I hope you did not trigger one of those Hey, I have a solution for you, hey, this is a non-commercial-list, go die flamewar - we had enough of those ;-) Best regards, Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tuesday 09 October 2007 14:32:38 Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm And your point, precisely, is what? Someone who has a criminal record can't be a technical authority? Someone can't have a criminal record without being a scumbag? Or perhaps that you prefer to write off those who can best your technical prowess by any means possible? My money's on the latter. -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime woes
I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for NAT-ed clients on different asterisk servers in an HA environment(Loadbalanced). Through packet sniffing I have observed that traffic is not being passed to the clients via the asterisk server they are registered to, hence breaking the call. Any insight on this would be great as the documentation on this subject is almost non-existent. Here's all the configs. sip.conf [general] svrlookup=yes displaysystemname=yes ;rtcachefriends=yes rtsavesysname=yes canreinvite=no externip=10.100.1.31 extensions.conf [internal] switch = Realtime/[EMAIL PROTECTED] res_mysql.conf [general] dbhost = 10.100.1.32 dbname = asterisk dbuser = asterisk dbpass = *** dbport = 3306 dbsock = /var/lib/mysql/mysql.sock extconfig.conf sipusers = mysql,asterisk,sip_users sippeers = mysql,asterisk,sip_users extensions = mysql,asterisk,extensions_table And heres the DB config mysql select * from extensions_table; ++--+---+--++---+ | id | context | exten | priority | app| appdata | ++--+---+--++---+ | 1 | internal | 111 |1 | Dial | SIP/tim | | 2 | internal | 111 |2 | Hangup | | | 3 | internal | 222 |1 | Dial | SIP/lance | | 4 | internal | 222 |2 | Hangup | | ++--+---+--++---+ 4 rows in set (0.00 sec) | id | name | host| nat | type | accountcode | amaflags | callgroup | callerid | cancallforward | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | insecure | language | mailbox | md5secret | deny | permit | mask | musiconhold | pickupgroup | qualify | regexten | restrictcid | rtptimeout | rtpholdtimeout | secret | setvar | disallow | allow | fullcontact | ipaddr | port | regserver | regseconds | username | ++---+-+-++-+--+---+--++-+--+---+--+--++--+--+-+---+--++--+-+-+-+--+-+++++--+-+-+--+---+-++--+ | 1 | lance | dynamic | yes | friend | NULL| NULL | NULL | lance| yes| no | internal | NULL | NULL | NULL | NULL | NULL | NULL | NULL| NULL | NULL | NULL | NULL | NULL| NULL| NULL| NULL | NULL| NULL | NULL | lance | NULL | all | g729;ilbc;gsm;ulaw;alaw | | 10.100.1.32 | 30988 | sanbox-mono | 1191962717 | lance| | 2 | tim | dynamic | no | friend | NULL| NULL | NULL | tim | yes| no | internal | NULL | NULL | NULL | NULL | NULL | NULL | NULL| NULL | NULL | NULL | NULL | NULL| NULL| NULL| NULL | NULL| NULL | NULL | tim| NULL | all | g729;ilbc;gsm;ulaw;alaw | | 10.100.1.108 | 64230 | kickstart | 1191963352 | tim | Note in the above example, lance is behind a NAT and tim is not. In this case tim cannot call lance but lance can call tim!. Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I don't see how this is relevant to the discussion. Zoa Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm On 10/9/07, *Brian West* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You apparently don't realize you're talking to. Thats ok, You keep working on it from your angle. We are evaluating when the time is right to implement this. We aren't doing this for Asterisk we are doing it for FreeSWITCH. /b On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote: Competition is a good thing. Let's say you fail or your implementation is not as robust as the other project or visa versa. Just as long as the hardware vendor is different, it should be a good thing. If it the same hardware vendor, then maybe you two should work together. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
I could not tell you in asterisknow but I use this feature with Polycom phones on all of my installs. It is very well documented in voip-info.org Do you have any problem with the Paging when there are say 20 phones in the page group? We have a IP601 that is used by the receptionist and has 2 side cars. We have to keep presence (Buddy List) enabled so the sidecar lights go on and off. However, about 1 out of 10 times the receptionist pages, her phone reboots. Polycom says it can't handle the traffic from the buddy list presence notifications. Have you seen this? Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Talk Web Applications with Asterisk
zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
On Tuesday 09 October 2007 14:20:33 Brian West wrote: I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. I'm not exactly sure that you're the right person to be taking offense at someone dragging a project's name through the mud. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy
softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Just use an 'invalid' extension to send the call to voicemail or something.
exten = i,1,Voicemail(u${INVALID_EXTEN})
I would *love* for it to be that simple, but I'm not doing this from an
IVR. Voip-Info says about the 'i' extension:
The 'i' extension only gets fired when there's a prompt or input been
made with 'background'. You can set up a 'exten = i,1...' to prompt for
wrong keypresses - insult the user and so on. So this wont work if
someone just dials somthing wrong.
--
Kyle Sexton
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Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.
On 18:04, Tue 09 Oct 07, Kyle Sexton wrote:
JR Richardson [EMAIL PROTECTED] writes:
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten = _,2,Answer()
exten = _,3,Wait(1)
exten = _,4,NoOp(sipregistration call - Name: ${CALLERID(name)} Num:
${CALLERID(num)})
exten = _,5,Macro(stdexten,${EXTEN},SIP/${EXTEN}) ; 913-563-77XX
This works really well for hard phones. They register, exist in the
sipregistration context and are dialed on whichever server they register
by DUNDi. I only started to run into problems when I had to deploy
softphones.
The softphones register when they are up and running, and the system
works as designed. But when they close their softphone, there's no way
for the system to know where the extension is, so the call dies. It
doesn't go to voicemail like I would like it to because that extension
never proceeds through my dialplan.
Looking for suggestions on getting around this so I can keep deploying
soft phones to agents in the field.
Just use an 'invalid' extension to send the call to voicemail or something.
exten = i,1,Voicemail(u${INVALID_EXTEN})
I would *love* for it to be that simple, but I'm not doing this from an
IVR. Voip-Info says about the 'i' extension:
The 'i' extension only gets fired when there's a prompt or input been
made with 'background'. You can set up a 'exten = i,1...' to prompt for
wrong keypresses - insult the user and so on. So this wont work if
someone just dials somthing wrong.
I did not follow this thread at all so please excuse me if
I'm redundant.
If you dont trust the i exten to handle things make sure
everything you want to handle is configured and put
something like this in you dialplan:
exten = _.,1,Goto(i,1)
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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Re: [asterisk-users] T-Mobile and WiFi Voip
On Tue, 9 Oct 2007 12:06:24 -0400, Jason Aarons \(US\) wrote: Will this work backwards? When I'm at home instead of my cell ringing have the home phone ring? Why would anyone give up revenue from minutes? Most won't...at least not for while. T-Mobile is the only offer available right now...simply because they're the only cellular carrier not also in some way in the land line business. It's a way for them to accellerate the trend of people dropping land lines in favour of only cellular service. If I could get my Asterisk server to mimic the wifi phone then it would be an ideal way of providing 911 and 411 service to my home. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, October 09, 2007 12:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T-Mobile and WiFi Voip Yep all the carriers are looking to offer 'voip' services sooner rather than later. Basically it uses the wifi point to access the mobile switching network. Cool part is you will soon be answering your Verizon home phone on your cell when you are 'within range' or your home network. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Sent: Tuesday, 9 October 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T-Mobile and WiFi Voip I had a friend yesterday showing me his new T-mobile blackberry with WiFi Voip.I could not believe it until I actually saw him making calls. There is no T-Mobile cell coverage at my house but he was able to simply access the WiFi router and make the call. It appears this VoIP offering is tightly integrated since you use the same phone number to make and receive calls over WiFi or Cell. Does anybody know if its SIP? I wanted to get some packet captures but he was in a hurry. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
Hello Gentleman Ladies , On Tue, 9 Oct 2007, Tilghman Lesher wrote: On Tuesday 09 October 2007 14:20:33 Brian West wrote: I'm number three on the dev team and not the soul person behind FreeSWITCH. Its very uncalled for. You are dragging our project thru the mud now also. Don't pass judgement on me. You sound quite childish and waste my time. Never judge a man till you walk a day in his shoes. I'm not exactly sure that you're the right person to be taking offense at someone dragging a project's name through the mud. Please , step back form the keyboard , take a deep breath . then maybe we can get on with the discussion of creating a driver under aterisk for a ds3 card . Tia , JimL -- +-+ | James W. Laferriere | System Techniques | Give me VMS | | NetworkEngineer | 663 Beaumont Blvd | Give me Linux | | [EMAIL PROTECTED] | Pacifica, CA. 94044 | only on AXP | +-+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DS3 Interface
I would recommend doing it on a 64bit platform for sure. Not sure Asterisk has very many linger issues on 64bit... I know I run it on 64bit without too much drama. /b On Oct 9, 2007, at 9:32 PM, Mr. James W. Laferriere wrote: Please , step back form the keyboard , take a deep breath . then maybe we can get on with the discussion of creating a driver under aterisk for a ds3 card . ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime woes
I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for NAT-ed clients on different asterisk servers in an HA environment(Loadbalanced). You need to do a local PBX lookup for the extension, if not present then do a DUNDi lookup to the peer PBX and pass the call over there. Also keep rtcachefriends=yes enabled so asterisk caches the sip registrations. You can also just dip the database and get the 'ipaddr' and 'port' info to call the extension directly. Give this a read if you haven't already. www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepa per.pdf Good luck. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libdundi?
Now the next question is why do no LGPL Dundi libs exist? /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4.11 function queue
i am configured asterisk-gui the Queue Extension Configuration but
configure and register into queue.conf :
[6]
fullname = Call Center
strategy = ringall
timeout = 5
wrapuptime = 5
autofill = yes
autopause = no
maxlen = 0
joinempty = no
leavewhenempty = no
reportholdtime = yes
musicclass = default
member = Agent/60010
member = Agent/60011
member = Agent/60014
but not register into agent.conf
is emply
extension.conf
exten = 6,1,Answer()
exten = 6,2,Queue(${EXTEN})
and users.conf with users.conf
[60011]
callwaiting = yes
cid_number = 60011
context = numberplan-custom-2
fullname = Roberto Bolivar Casa
hasagent = yes
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
deletevoicemail = no
host = dynamic
mailbox = 60011
secret = 123654
threewaycalling = yes
vmsecret = 1234
registeriax = no
registersip = yes
canreinvite = no
nat = yes
dtmfmode = rfc2833
disallow = all
allow = all
the client Xlite call to 6 and the enter to musicold and not pass to the
agent ...
how to the problem ???
and the asterisk-gui not can edit the Call Queues rules and show message:
You can not edit the selected entry from here. Please click on the 'Users'
panel to edit the selected entry
how to problem , read to the loging agent ??? but no loging.
THANKS!
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