Re: [Asterisk-Users] * 1.2.4 1.2.6: Ringing anamoly
Ronald Lewis wrote: I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned Broadvoice issue. For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence. Any ideas? Just out of curiosity did you happen to put an Answer() before playing audio or ringing? I use BroadVoice also and I used to have the exact same problem but putting Answer() as the first step in the context before playing my menu solved the problem. -Chris -- Chris Shaw IT Manager Precision Pump, Inc 150 N Main St Banks, OR 97106 Phone: 503-324-2361 Fax: 503-324-2203 E-Mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring Using Flash
Will Szopko wrote:
Greetings.
I am attempting to configure a system based on Asterisk 1.2.3 to be used
as a backup should our aging voice mail/auto attendant system fail, which
seems increasingly likely given its advanced years. The first part of this
task is getting the auto attendant feature to work correctly, which I
would have figured to be relatively easy. I have successfully built a menu
structure, but cannot get Asterisk to transfer calls back to the legacy
PBX (Fujitsu F9600). In essence, all I require Asterisk to do is:
1) read the extension digits entered by a caller;
2) flash the line [Flash()];
3) dial the extension using DTMF [SendDTMF(${EXTEN})]; and,
4) hang up [Hangup()].
Unfortunately, I've not been able to make this work and was hoping someone
might tell me where I'm going wrong. The problem appears to be in the
flash portion of the above procedure.
Asterisk Server Setup
-- -
- Ubuntu Linux 5.10 (Breezy Badger) for AMD64
- Linux 2.6.15 kernel (custom-built)
- Asterisk 1.2.3 (built from source)
- Zaptel 1.2.2 (built from source)
- Digium TDM2402E (8 FXO ports)
Legacy PBX Hookup
-- --- --
The Asterisk server is connected to our Fujitsu F9600 via 4 analog
connections with the 9600 providing dial tone.
What I Want to Happen
- -- --
1) Call comes into legacy PBX.
2) PBX transfers call to Asterisk.
3) Asterisk goes through greeting and offers to take an extension to which
to transfer.
4) Caller enters transfer.
5) Asterisk transfers the call back to the PBX using the steps described
above.
Asterisk Configuration
-
/etc/zaptel.conf
fxsks=1-4
loadzone=us
defaultzone=us
/etc/asterisk/zapata.conf
[trunkgroups]
[channels]
; hardware channels
; default
usecallerid=no
hidecallerid=yes
callwaiting=no
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echotraining=yes
busydetect=yes
callprogress=no
; define channels
context=greeting
signalling=fxs_ks
channel = 1-4
/etc/asterisk/extensions.conf
exten = _[45]XXX,1,Flash()
exten = _[45]XXX,n,SendDTMF(${EXTEN})
exten = _[45]XXX,n,Hangup()
What Happens
---
-- Executing Flash(Zap/4-1, ) in new stack
Jan 26 16:10:17 WARNING[4564]: chan_zap.c:3907 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 4
-- Flashed channel Zap/4-1
-- Executing SendDTMF(Zap/4-1, 4424) in new stack
-- Executing Hangup(Zap/4-1, Zap/4-1) in new stack
Upon executing the Hangup command the phone goes dead without the transfer
having been made. The one odd thing here is the warning about the strange
state 6 on channel 4.
Other Things I've Tried
- -- -
1) I've tried a phone plugged directly into one of the lines on the PBX,
did a flash on the phone, and successfully transferred a call with no
problems.
2) I swapped out the TDM2400 and tried a TDM400. It does the same thing as
above, but without the strange state warning.
3) I tried Asterisk 1.0.10. It does the same thing.
If anyone has any ideas of what may be going on here, I'd very much
appreciate some assistance. As I'm learning about Asterisk I am finding a
lot to like, but am getting frustrated that I cannot make this work.
Thanks for your help.
- Will
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Don't get discouraged!
It looks like you've got the right idea. What might be happening is that
Asterisk isn't waiting long enough for the PBX/KSU to respond to the
flash. What you might try is inserting a wait(1) in between your flash()
and your SendDTMF() like this...
exten = _[45]XXX,1,Flash()
exten = _[46]XXX,2,Wait(1)
exten = _[45]XXX,3,SendDTMF(${EXTEN})
exten = _[45]XXX,4,Hangup()
If that doesn't work try chaing wait(1) to wait(2).
The other thing that might be happening is that the flash is too short
for your PBX/KSU to recognize it. If your PBX/KSU supports it, you could
try changing the flash timing through system programming. Nortel
NorStars call it Link Time other systems call it Reach Through. Try
setting the timing between 400 and 600 ms and also keep the Wait(1) in
the dialplan to give your PBX/KSU enough time to handle the flash.
Hope this helps you!
-Chris
--
Chris Shaw
IT Manager
Precision Pump, Inc
150 N Main St
Banks, OR 97106
Phone: 503-324-2361
Fax: 503-324-2203
E-Mail: [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardware vs. Network Inputs
Michael, Doing an All-Network setup is completely doable but there are many factors to consider. First of all, I didn't see any mention of how many connections it takes before Asterisk starts having difficulty with DTMF. You mentioned that the computer is directly connected to a T1, is it the only computer using the T1 or are there others? Also what kind of network is it? Do you have a good SLA? What kind of packet loss do you experience on average? What is your ping time to the Broadvoice proxy that you're using? Are you using any kind of QoS? Remember that Broadvoice only uses G.711u/a so with RTP + UDP + IP overhead you're looking at ~85kbit/s so at around 9-10 concurrent calls you're going to be pushing it a bit with 900Kbit available bandwidth. You might try turning the SIP RelaxDTMF setting on, that may help, also if you don't have and are not planning on getting any Zaptel hardware, consider using Ztdummy or ZapRTC as an RTP timing source. I know that on the wiki it says that they are really only useful for MoH or MeetME but I've found it to help greatly with audio quality and Asterisk's DTMF detection. YMMV. Good Luck! -Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice
Working fine here in the Northwest. Actually I haven't had a single problem with them since the dreaded Global Crossing fiasco... -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] McLeod Integrated T1 - no PRI?
**Snip** pbx*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 Take a second look at your status. It says that the D-Channel is down... No D-Channel, No PRI Signalling. Tell McCleod to bring up the D-Channel so their switch can talk to yours... -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_valetparking / parking in general
Does anyone have Music-On-Hold and valet parking, or regular parking working together? No matter how I configure it, I cannot get moh to continue to play after I park a call using either valet parking or regular parking. The only thing I can think of is that I might need to use # transfer instead of sip native transfer? Shouldn't this just work? If needed I can post the config for one of the 50 or so different ways I've tried to make this work so far. Sounds like your MOH is not working in general. It works for me in both Asterisk native # transfer and SIP Native REFER transfer... As soon as the transfer begins, MOH should start on the channel (indicated in the console if your verbosity is high enough). Can you provide a console output? A debug output? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream BT101 Attended Transfers
I've asked Grandstream tech support about attended transfer. They told me that in about a month there will be available a firmware upgrade that supports attended transfer natively. I never heard this, SWEET! You're not kidding right? This is something the phone REALLY needs. Now if they could just get the conference button to be something other than just a decoration... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Medium volume 100% SIP/IAX PBX.
Hello, Hi Does anyone have experience with a %100 all VoIP * setup? I imagine an office with 50 extensions or so, a full T1 connected to a decent ISP and an account with NuFone (IAX2 trunking, G729a). I would need to support at least 25 simultaneous outgoing (7960G - Asterisk - NuFone - PSTN) phone calls. I would probably keep four or so analog lines for local calls, 411, 911, etc. Does anyone have any personal experience with such a setup? Would you put YOUR business on it? I would be willing to throw as much money as needed at hardware, dual Xeons, etc, if it would at least be possible. * should have no problem keeping up with a setup like that, and VoIP is certainly capable However... What most interests you? Is it cost savings or audio quality. If it's cost savings, you could push 25 calls through a T1 using GSM encoding, but it would not sound quite the same as a regular line. If you use G.711 (mu/A-Law) then you would get toll quality audio but only be able to push about 17 calls through at once... Also you must remember that the current RTP implementation in Asterisk is... somewhat... lacking, and with a 100% VoIP setup you will need a timing source like ztdummy (which requires a UHCI USB controller) or ZapRTC. Or if you're using linux 2.6, I don't think you need anything as the internal timer resolution is precise enough... Our company is thinking of deploying a setup like this but a bit smaller, only 12 extensions and at most 8-9 simultaneous calls. I would certainly recommend a setup like this, it's a huge cost savings. I would also do plenty of homework and figure out how to do it before actually committing to it. Maybe even do a parallel setup where you have some POTS lines as a backup. I would also use some kind of failover where your IAX provider can forward your incoming calls to another IAX provider's number or a POTS number during downtimes... Thanks for your responses. You're welcome -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Medium volume 100% SIP/IAX PBX.
I know of several that are working fine. We've got our small business on it too, but we're doing BV via a DSL; we have much lower call volume requirements. You say you use BroadVoice? How are you dealing with the voicemail issue? How about multiple simultaneous calls, are you paying for multiple plans or do they allow that on their business plan? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting
The other issue is that call waiting does not appear to work. The way I'm expecting it to work with Asterisk is to send the second call to me - I'm using SetGroup and CheckGroup within Asterisk to limit my calls to two at a time total. However, if I'm on a phone call (incoming or outgoing), Broadvoice transfers a second call to a person you are calling is busy message -- I don't see any additional SIP traffic to the Asterisk box. You must have call waiting turned off on your comm pilot control panel, go to www.broadvoice.com and log into your control panel and make sure call waiting is turned on. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Outbound Proxy
I've used it and it works great! I think it's vital that chan_sip include outbound proxy support. * is not only acting as a PBX and a telephony gateway but also to the termination provider it acts as a SIP UA and needs to have all of the features that a SIP UA would have including outbound proxy support, DNS lookups and RFC-Compliant responses... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicebox
Yup, i did :)
I found a lot about setting up my voicemail now.
But i couldn`t find much how you collect/call/retreive the recorded mail.
Any ideas/links about that?!
Thanks, Mario
ok, that would be a dialplan issue. You need to do something like this in
your dialplan.
[mycontext] ; The context that your phones live in
exten = 770,1,VoiceMailMain() ; I didn't use _ because it's an exact match.
- OR -
if you have a phone that has a programmable voicemail button this is a neat
trick...
This assumes your mailbox number is the same as your extension number as it
is with most PBXes...
[mycontext]; The context that your phones live in
exten = _770XXX,1,GotoIf($[${EXTEN:3} = ${CALLERIDNUM}]?2:100
exten = _770XXX,2,VoiceMailMain(s${EXTEN:3}); the 's' means skip login
exten = _770XXX,100,Congestion()
This is kinda handy for my household, we use GrandStreams and they have a
programmable VoiceMail button, I simply program the button to dial
770exten and voila!
Some useful reading material...
http://www.voip-info.org/wiki-Asterisk+cmd+voicemailmain
http://www.voip-info.org/wiki-Asterisk+cmd+gotoif
http://www.voip-info.org/wiki-Asterisk+conf+extensions.conf
http://www.voip-info.org/wiki-Asterisk+variables
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3-way calling
That works exactly as expected on Zap interfaces. For VoIP devices it's TOTALLY handled by the phone. If you're extremely lucky :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with length of voicemail
-Original Message--- Yes, I *am* using BROADVOICE, thanks for responses. Sure enough. If I dial in via my Washington number (ipkall), I don't have the problem. Interesting. Well, BV has a very good tech that seems to be very familiar with Asterisk. I'll see if he has any ideas how to deal with the issue. Sorry I didn't catch the earlier thread -Begin Reply-- Myself and several others have had this problem. (Anyone using an ITSP who uses BroadWorks I imagine). My current theory is that BroadWorks requires some acknowledgement that the sending side, Asterisk, is still there. Right now because * has no CNG (Comfort Noise Generation) or DTX (Discontinuous TX) support, it does not send anything back to the receiver until recording is finished... BroadWorks takes this as a sign that * has lost the connection and tears down the connection... (You can see this in *, it says User Hung Up). I did mean to say BroadWorks, that's the brand of VoIP switch that BroadVoice uses. I have a strong suspicion as stated earlier that any provider that uses BroadWorks WILL have this same problem... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
I haven't forgotten!! :) Just an update on my progress, I had to work on Saturday and the family went on a trip for labor day so I didn't get a lot of time to work on the Wiki. This week I'm planning on getting some examples put together, I also wrote a shell script that can be used at boot time to set up QoS on your Linux Bridge or NAT Router... This and much more will soon be available on the Wiki! Keep checking the Wiki, I'm hoping to get this done sometime this week! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tormenta Asterisk
Wow that's an old card! I believe the TORISA (Tormenta ISA) driver that comes with the Zaptel library will work... You'll be using up ALL your ISA bus bandwidth with just that one card however... - Original Message - From: Sergio Galeotti To: [EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 12:13 PM Subject: [Asterisk-Users] Tormenta Asterisk Hello: I have a "Tormenta ISA Card, Rev. A" and wanted to know if somebody knows like integrating it with Asterisk. It would thank for any information on the matter. Thanks ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/ojitterbuffer enabled?
- Original Message - From: Kris Boutilier [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 4:26 PM Subject: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/ojitterbuffer enabled? I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from 20 tos=lowdelay notransfer=no trunk=no All calls are running as GSM, even though g.729 is also an 'allowed' codec (w/5 licenses installed). During an average call 'iax2 show channels' provides: Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 10.0.40.140 astpbx-woo 2/2 5/6 00040ms 0036ms ms GSM If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP doesn't just disconnect sockets unless it recieves a RESET or a FINISHED or there's a timeout (usually like 5 minutes or more depending on your TCP/IP stack). Needless to say that to disconnect a TCP connection, that would have to be one hell of a lag spike... * must be actively disconnecting the connection I've heard the jitter buffer is a bit buggy, have you tried turning it off completely? Hell even SIP won't just disconnect you unless your UA tells it to do so... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe without ZAP?
- Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 4:39 PM Subject: RE: [Asterisk-Users] MeetMe without ZAP? Matthew Boehm wrote: Since I am using a SMP machine without USB ports does that mean I am fuX0red and can't run MeetMe at all? You can try the zaprtc (search for a link), or go out to Staples/OfficeDepot/BestBuy and pick up a PCI USB adapter. It must be a UHCI USB adapter though and that's not usually written on the box anywhere! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/ojitterbuffer enabled?
Is 'jitterbuffer=no' not sufficent to clobber that function? Yes, sorry I didn't see that part... Hmm And IAX2 debug on doesn't give you any hints at all? like who hung up first, etc... ? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sorry, Newbie here
I think one of the greatest things about * is that not only do you get the most flexible PBX I've ever worked with, but it also can act as a IP gateway for much less than traditional hardware IP gateways (a. la. Cisco/Mediatrix/etc...). You can use it to extend an existing PBX and save thousands per month by terminating your PSTN calls via IP... -Chris - Original Message - From: Jason Kawakami [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 03, 2004 9:24 AM Subject: [Asterisk-Users] Re: Sorry, Newbie here - Original Message - Subject: [Asterisk-Users] Sorry, Newbie here To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I never heard of Asterisk before today, but from what i'm looking at on the website and hearing, it sounds pretty incredibly. If I understand correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux Server, and a couple IP phones or Netmeeting on a few workstations, and of course, Asterisk which is free; I call have a small call center. This can't be? I was looking at tens of thousands for a Cisco solution. Any comments or insight is welcome. after working the telecom industry for the past 10 years i can tell you to believe it. your statement is absolutely true dont kid yourself though, * has some gotchas especially in call center functionality, and * requires learning from scratch how open source software developers interpreted what hardware engineers have done for the past 30 years. if you have experience in implementing open source solutions and some telephony background you can build just about anything you want to do with a telephone and a computer with *. usually there is a trade off in cost (read capital expenditure) and installation and maint of these solutions. i would suggest to you contacting a consultant (check the listings on voip-info.org) and contact someone near you about your requirements. or do what we all did and download the software from CVS and dive in. welcome to the brave new world Jason Kawakami www.optellabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digitnetworks card issues?
The T400P (and E400P) are clones of the Zapata Tormenta II, and anyone can download the artwork to build and sell their own version. If the owners of the Zapata Telephony project didn't want people to use their designs then they would not have released them under the GPL and published them on a public website. Last time I looked on there I think they even published the gerber files so you could feed them into a CAM -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] digitnetworks card issues?
Lol... This never clicked before... It's called Zapata Tormenta (Shoe Storm)... Like a bunch of women at a shoe sale I guess... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
I'd be more than happy to send you some info off-list on how to do this in Linux... It's much cheaper and more flexible than a low-end hardware solution... -Chris - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 03, 2004 2:30 AM Subject: [Asterisk-Users] Lower cost router suitable for VOIP ? Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we use to give voice packets higher priority. I'm avare of VOIP routers, but they are pricey. Can some of common routers help, or maybe implementing router on another simple Linux box? Any advice, pointers to more info ? How to trace network and debug Asterisk in convenient way ? Thanks in advance, Robert Rozman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VoIP?
Chris, I believe it would be nice to send the info also to the list. So others would be able to benefit as well. You've got at least 2 people interested :) Marconi. Ok, I just wasn't sure because it's more of an 802.3Q/Routing issue rather than an * issue, but if everyone's cool with it I sure will... I'm not the only one who knows this stuff and I might not even be doing it the best way but it works for me and I'm using it with *... - Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
How about the Wiki? :-) I think I'm gonna have to because it would be too long to e-mail! I can give you guys the short version though... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
Well.. ok... here goes the Short version, I will be adding examples and explanations to the wiki when I get off work... :-) Bear in mind this is what I do, change it to fit your situation... I'm on a cable modem which everyone knows just BLOWS for latency, also it's an external one so you can't control the buffering... but I've been able to use Linux QoS to make it near toll-quality with the occasional jitter during heavy downloading... I have 3Mbit download speed and an abysmal 256kbit upload speed... Needless to say that upload is a problem when shared between 6 machines... Everything that you do requires sending SYN/ACK packets and such which destroys upstream band... Unless you use QoS these packets will just be thrown at the interface willy-nilly with no regard for speed and time... There are 2 ways that I know of to do this and because of the topology of my network I actually use BOTH methods so I know it works very well! The first is to use the linux bridging code included in the 2.4.X and 2.6.X series kernels and the bridge-firewalling code included with the ebtables project (http://ebtables.sourceforge.net) to create a Layer-2 ethernet switch with QoS support. I use ebtables and it's packet marking target to mark packets that are received from my LAN and are destined to be bridged to my WAN interface hooked into the cable modem. Then I create QoS filters based on those marks... Using ebtables also allows you to mark packets based on their destination MAC whereas iptables does not... Bear in mind that this is a software switch not a hardware switch so it can pass packets at wire speed but some network drivers are horribly broken and slow (rtl8139, 3c90x, eepro100, etc..) and also when you open a lot of TCP sockets simultaneously it uses a lot of memory and CPU... This works beautifully and to the end users and applications it's completely transparent! The second way is to simply use IPTABLES and NAT to create a NAT router. In this scenario you're just using iptables' connection tracking code to do NAT/MASQUERADING (like in the good ol' IPCHAINS days of 2.2.x or the IPFWADM days of 2.1.x!). In this situation packet marking is done in the MANGLE table, in the FORWARDING chain... For those of you who feel brave/foolish enough to use the U32 packet matching code instead of marking the packets, that will work for the NAT router but not in the way you would expect for the bridge because it works at layer 2... If you already have a router like a LinkSys or a Dlink that doesn't support Qos, don't worry! I would suggest using the Linux bridge code and placing a linux box between your LAN and the router. That way you can implement QoS and strong firewalling based on IPTABLES to your hardware router! Again, I'm going to be posting examples of my setup on the wiki. Also I've written an init-style script for ebtables and am currently working on an ifup style script for the bridge device. That one's tricky because the bridge code doesn't pass packets for 30 seconds while it's Learning. Also the bridge device is traditionally not assigned an IP address... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
I'll be sure and include all of the important links I've used as well... Bear in mind that this will only help YOUR network, if your ISP's link to the rest of the world sucks then you still won't get the desired results... but with a little bit of network grooming, I think most people will be able to get the results they want even on DSL/Cable! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
Also I'm not trying to say Linux is better than FreeBSD, I know FreeBSD has a similar implementation, unfortunately I'm not familiar enough with BSD's Bridging/Firewalling/Routing implementation, anyone with BSD experience who wants to add to this feel free! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 03, 2004 3:06 PM Subject: Re: [Asterisk-Users] Lower cost router suitable for VOIP ? On Friday 03 September 2004 17:48, Chris Shaw wrote: based on their destination MAC whereas iptables does not... Bear in mind that this is a software switch not a hardware switch so it can pass packets at wire speed but some network drivers are horribly broken and slow (rtl8139, 3c90x, eepro100, etc..) and also when you open a lot of TCP sockets simultaneously it uses a lot of memory and CPU... This works beautifully and to the end users and applications it's completely transparent! eepro100 is horribly broken? I can saturate the link without breaking a sweat -- not in CPU nor memory. The drivers have gotten much better, but yes, up until about 2.4.22 it used to hard lock my server every 24 hours or so under heavy packet loads... Remember what I said about it being a software bridge... Intel cards in general are not known for being CPU-Friendly... Really the tulip based cards are the fastest I've seen, I know they're kinda cheap boards usually but they scream performance-wise... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
Generally speaking, the bulk of data flowing in any end-user pipe is contained in TCP and that can be rate limited on the receiving side. UDP traffic you're more or less out of luck with unless the ISP supports ECN which many do not. So really the key to VOIP on consumer grade connections simply not to fill your pipe, since you have no control over what is prioritised. That's basically what I'm doing... I'll post examples but that's about all you can do... It's really up to the ISP, but all I'm saying is that you can have a pretty decent setup without Carrier-Grade SLAs, you CAN do it over pretty much any broadband connection within reason of course... Also jitter can be a problem if the ISP is using RED... RED really falls short for UDP, it was designed with TCP's backoff algoritms in mind. RED like many other QoS schedulers works by dropping packets... this is not good for VoIP... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
Actually to be fair I think it was more a combination of the NIC driver and the connection tracking code at the time that lead to the lockups... With connection tracking and bridging, the firewall tracks EVERY connection, not just NAT so it can use a lot of CPU/Memory... But the Intel drivers can be a bit slow and CPU-intensive, there's been lots of discussion about that... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Putting a call on hold
How do I put a call on hold? If i press # the music on hold plays to the other person, but asterisk asks for a number to transfer... I don't want to transfer, I simply want to put the person on hold, so he/she can hear the music while I do something, then get them off hold. Is it possible? The scenario: The person calls me from a SIP phone, and I receive the call in a regular PSTN phone, from the FXO. I don't want any of the calls to get disconnected. This is normally implemented by your analog phone with the Hold button. If you don't have a hold button you can use call parking that way you can pick the call back up from any phone... To park, simply press # and then when Allison says Transfer? you dial 700. She will then read off the digits of the parking extension where you can pick the call back up. You need to have the [parkedcalls] context from parking.conf included in your current context in extensions.conf for this to work properly. e.g. [mycontext] include = outbound-local include = outbound-tollfree include = parkedcalls -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
That's not my experience. I don't remember any eepro100 driver lock-ups, and I have at least 1000 card-years of experience with them, ranging back from 2.0.x to 2.6.x. We replaced all of our 3com cards due to driver problems (circa 1998), but the Intel cards Just Worked. We never noticed a CPU load problem, but we were only rarely concerned with CPU load, anyway. Ok Way OT, I didn't mean to get into a religious debate, I like the Intel cards, I have several of them and recommend them to my friends, etc... Be that as it may... This was using these cards in a software bridge... significantly more traffic than an ordinary end-to-end connection... Packets destined for MANY different PCs are being passed through the card... It may have been a combination of the bridging code and the NIC drivers that lead to the instability problems I experienced... I've always been nervous about Tulip clones. I have a half-dozen 21143 boards at home that are great. The tulip cards are awesome for the simple fact that they're hella old... (yes that's the scientific reason!) The tulip design goes back to the old DEC 21040 chips of the early-mid nineties (ahhh the good ol' days!) There has been a lot of time to play with these chipsets and they are well documented so pretty much all of their functions work well and there are no surprises The problem with Intel/3Com/et. al. is that the open source drivers either have to be reverse-engineered or the company has to be open-source minded enough to share information... There are many undocumented features in these kinds of drivers that just kinda work(tm)... Again... I have no affiliation with one or the other, no religious prefrences, no nothing... This is just what I've found using the cards... YMMV! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why do i get this message emailed to meeverytime i post?
The proper thing for you to do is find or spam a real admin account at bembang.com to fix their broken software and admonish the user for not checking and cleaning mail more often. They don't care... I've sent several messages to the postmaster and they go unanswered... Obviously a bunch of idiots... can't we just forcibly Unsubscribe this person? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Configuration for Mixed Voice Data
I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what encapsulation they use it's still HDLC. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15 is-slave = off DS0 Provisioning: analog-begin = 1 analog-end = 16 data-begin = 17 data-end = 24 alignment = same -zaptel.conf- span=0,0,0,esf,b8zs ; Set up a span number 0 with the provider as the timing source, an LBO of 0-133ft with esf framing and b8zs coding. fxsks=1-16; basically if you're not on a PRI then you're on a channel bank... You need to know what kind of signalling your provider uses, ; is it LoopStart, GroundStart, EM or KewlStart... nethdlc=17-24; Combine channels 17-24 into data for the Linux HDLC layer... -zapata.conf- signalling=fxs_ks context=yourcontext channel = 1-16 From here on out, it's not an * issue, it's a Linux HDLC Layer issue which is beyond the scope of this list... You have enough information to get it working though, from your Vina you can see that your DLCI is 100, you'll need the gateway address of the router on the other end and DNS information. Also whether or not they're using PPP encapsulation, all of this is configured with the sethdlc program and also /etc/resolv.conf and /etc/sysconfig/network and the ifup scripts if you're using RedHat. That should work... I might have the FXO/FXS thing reversed I'm always doing that, but if it doesn't work, reverse them and it should... Like I mentioned in my comment you need to see what kind of signalling your provider is using, is it GroundStart, LoopStart, EM/EM Wink or KewlStart... I haven't done this but I am thinking about switching to a setup much like this so if you have success/failure let me know! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Configuration for Mixed Voice Data
Ok, firstly HDLC is just a Layer 2 protocol like IP, so no matter what encapsulation they use it's still HDLC. I meant Ethernet/ARP IP is at layer 3 DUH... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] latest CVS build won't load
Make sure you delete your /usr/lib/asterisk directory before installing a new CVS copy... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] latest CVS build won't load
Hmmm... Well what that means is that the code is using pthread_create() instead of ast_pthread_create(), it's not a major thing, all you would have to do is go through all the affected modules and replace pthread_create with ast_pthread_create, but this should probably be fixed in CVS too! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
This is nothing to do with SIP. It is an RTP issue, common to everything which uses RTP - SIP and H.323 included. I have been reading the RFCs and I'm a bit more familiar with how it works now although the algorithms are a bit over my head. I am somewhat new to RTP/VoIP, but I have a strong telecom/networking background so it makes things a bit easier to understand since they share a lot of common features.. I just thought from the post mentioning only IAX2 and some of the other codecs that SIP et. al. would be ignored... Sending no packets is perfectly valid, and normal, in RTP. If the receiving end takes no packets (other than, perhaps, an extremely long silence) as a disconnect it does not comply with the RTP spec. DTX is much despised, and CNG only slightly better. They just sound good (pun intende) on paper. While I realize that hanging up on silence is not a desired behavior, unfortunately lots of things are out of spec... Look at Cisco's POE implementation for example, it's completely reversed from 802.3af specs... If * had at least some kind of continuous CNG capability it would help in these situations... Silence should be acceptable and even desired because it saves bandwidth, but apparently some people (and switches) find it uncomfortable... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-users] PLC (Packet loss cancel) questions
I have been reading the RFCs and I'm a bit more familiar with how it works now although the algorithms are a bit over my head. I am somewhat new to RTP/VoIP, but I have a strong telecom/networking background so it makes things a bit easier to understand since they share a lot of common features.. I just thought from the post mentioning only IAX2 and some of the other codecs that SIP et. al. would be ignored... OOPS I meant... * protocols that SIP et. al. would be ignored... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit the length of extensions
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten = you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to use 's'.
Using your example, your dialplan should look something like this...
[incoming]
exten = 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten = 9543340726,2,setcidname(Blocked)
exten = 9543340726,3,setcidnum(00)
exten = 9543340726,4,Goto(companyname,beginmenu,1)
[companyname]
exten = beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten = beginmenu,2,Wait,1
exten = beginmenu,3,Answer() ; Answer the channel!
exten = beginmenu,4,Background(company-main)
exten = beginmenu,5,Background(ifyouknow)
exten = beginmenu,6,Goto(company_mainmenu,s,1)
exten = 502,1,Dial(SIP/whoever1SIP/whoever2sip/whoever3,30,m)
exten = 507,1,Dial(SIP/daveSIP/jimSIP/lisa,30,m)
...
[company_mainmenu]
exten = s,1,Background(company-nav1)
exten = 1,1,Goto(company_sales,s,1) ; Sales
exten = 2,1,Goto(companyname,502,1) ; Accounting
exten = 3,1,Goto(companyname,508,1) ; Customer Care
exten = 4,1,Goto(companyname,507,1) ; Technical Support
exten = 5,1,Goto(companyname,202,1) ; Human Resources
exten = 6,1,Goto(companyname,202,1) ; Provisioning
exten = 7,1,Goto(companyname,214,1) ; Marketing
exten = 0,1,Goto(companyname,210,1) ; Operator
...
Instead of jumping back and forth like this, I'd use macros to try and
condense the dialplan a bit...
I can help you more with this if you'd like...
Then for people inside the company there's this...
[outbound-local]
exten = _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,t) ; for
7-digit dialing
exten = _91800NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91888NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91877NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten = _91866NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-ld]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-international]
exten = _9011.,Dial(SIP${EXTEN:[EMAIL PROTECTED],60,T)
[office]
include = outbound-local
include = outbound-ld
include = outbound-international
exten = _[1-5]XX,1,Dial(SIP/${EXTEN},25,tT) ; This is assuming they're all
SIP, you can use $DIALSTATUS to continue checking ZAP,MGCP,ETC...
and so on...
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple lines with SIP like MGCP?
The HT486 is a single-line device with a PSTN pass-thru. The only multiline IADs I know of are the SIPURAs and the Cisco ATA-186... What you do is you create 2 contexts, 1 for each line of the device and you set the host name to the IP address (or host name if applicable) of the IAD. Set the username of each context to the line's respective extension in Asterisk. Then in the web setup for the IAD, there should be a place to put the username for each line as well as the password... I have not tried this but it should work, SIP is not IP/MAC based it's more like SMTP, it's user based... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P No D-channels
Zaptel.conf sets t100p to be the primary sync source for the only span, as suggested by many Asterisk users. I'm trying to understand so please bear with me... The T100P is connected directly to the Mitel? Or to the Telco through a T1? What I mean is are calls coming into the Mitel from the telco and then from there going into * or are calls going into * first and then being fed into the Mitel? If your T100P is connected to the telco then the clocking source should be the telco as their clocks are going to be a LOT more accurate than your PC's interrupt timers... If your T100P is connected to the Mitel, then you've got it right... Just checking, I wasn't sure from your description... Occasional interrupt misses are pretty normal although in a perfect world with a good mobo they should not happen at all... If you're seeing multiple misses per second (e.g. everytime you do cat /proc/interrupts you see more) then there's a problem... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P No D-channels
H I guess from a troubleshooting standpoint to try and pinpoint the problem what I would do is remove all cards from the system and then only replace the cards that are absolutely necessary like your SCSI card and your Video card and of course the T100P and then check /proc/interrupts to see if you're having any more MISses... Also are you getting interrupt ERRs as well? Is APIC enabled for your board? If it is, you'll see things like IO-APIC-edge or IO-APIC-level in your /proc/interrupts. If not, you'll see XT-PIC for all interrupts... If you can't get APIC turned on you might try upgrading your kernel, your motherboard/bios may be blacklisted in that particular kernel... You definitely DO NOT want to share interrupts on the T100P unless it's a low-interrupt device like a USB controller or your video card. You definitely don't want to share with say a NIC or a SCSI controller... If it is sharing with one of those, try shuffling the cards around in different slots and make sure that your T100P isn't in slot 5. Slot 5 is usually shared with Slot 1 if they're on the same bus... Of course you've probably already tried all of this but just in case you haven't... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why is it called 'Comedian Mail?
I've wondered that myself... obviously the writer has a sense of humor! :) I like the sound of Digium Mail, it sounds cool... - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:30 PM Subject: RE: [Asterisk-Users] Why is it called 'Comedian Mail? Kris Boutilier [EMAIL PROTECTED] wrote: Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Perhaps it was written by someone with a red nose, oversized shoes and a custard pie. I don't know either. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can asterisk detect BUSY signal?
Lol reverse hold! I can't see that working ever though, I tried it once and the agent at the other end hung up on me... I had to wait another hour in the queue... - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:53 PM Subject: Re: [Asterisk-Users] Can asterisk detect BUSY signal? On Tuesday 31 August 2004 17:36, Kevin Walsh wrote: Spam-dialling should be made illegal. I, for one, wouldn't spend two seconds adding features to support this sort of usage. I can think of at least one legitimate use for this -- reverse spam dialling, or at least real person detection. I hate sitting in hold queues and my usual method is to put the phone on speaker and listen to Muzak while I wait. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
- Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete solution... As much as we all hate it's complexity and wish that everything would speak IAX (I know I do) a large number of devices support (and will be supporting) SIP, making it equally as important as IAX2 in using * as a complete telephony solution... DTX Support: Sending a single CN packet (in IAX2, this should probably sent reliably) would probably be good. I second, third and fourth this one as does anyone who's tried to use BroadVoice with Voicemail... Currently when * is not making any noise (e.g. recording) absolutely NO packets are sent back to the proxy... A lot of proxies take this as a sign that the far end has disconnected... Including BroadWorks! But they do recognize small CN packets as a sign that the SIP device (Asterisk) is still there... PLC I think is somewhat implemented already in codecs that support it, but I could be wrong, I remember seeing mention of it in the code... This would be SO helpful!!! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Nevermind, DUH, I was reading it wrong, it states that they DO NOT contain CNG algorithms, it describes a way to send CNG on codecs that do not contain CNG algorithms natively... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
The way I interact with BroadVoice though isn't officially sanctioned, I didn't prefer to use their Asterisk Only SIP gateway, in which they charge you 3.2 cents a minute (or whatever) when you exceed the first line. Where did you get this info? I have been using broadvoice for 2 months now and have never heard of this? I have heard of the 3.2cents a minute thing, but have never experienced it myself, I occasionally have calls on more than one line when others are using the phone and I don't know it... ok.. could we add a 'hunt group' to * and roll incoming calls over to several extensions? This totally defeats the purpose of VoIP This is going back to circuit-switched mentality... Remember that in VoIP a Line is just a username assigned to you by an ITSP, it can be a name or a number... You don't need rollover because it's just a connection like someone sending an E-Mail to your SMTP server... I realize what you mean, getting several accounts and Rolling them over so you can have multiple call appearances, but this breaks the whole idea of a pure VoIP setup... At $20-30 a month, you might as well use a TDM400P and add a second PSTN line, there are plans out there where when you sign up for a 2nd line you get unlimited long distance... My 0.0002 -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice User hung up on voicemail
Yep... It's BroadVoice's problem, not *'s... When * is recording, be it voicemail or the record() application, * does not transmit a single packet back to BroadVoice (Confirmed by ethereal and TCPDump) After 30 seconds the BroadVoice switch will disconnect the call believing that it's a far-end disconnect... I think that once CNG is implemented in *, this problem should be fixed, but until then, you get 30 seconds of recording... period :( -Chris - Original Message - From: Kevin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 27, 2004 8:20 AM Subject: [Asterisk-Users] Broadvoice User hung up on voicemail After a call is sent to voicemail on an inbound connection from Broadvoice, the call is hung up in the middle of recording a voice mail after about 30 or so seconds. I get an error User hung up. If I answer the call and not have it go to voicemail, the call will stay connected. This only seems to happen on the Broadvoice connection and voicemail. Is anyone experiencing this issue or able to resolve? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there any graphic designers on this list?
ooohhh I'll take a crack at it! sounds like fun! :) (B- Original Message - (BFrom: "Sunrise Ltd" [EMAIL PROTECTED] (BTo: "astusr" [EMAIL PROTECTED] (BSent: Friday, August 27, 2004 8:47 AM (BSubject: [Asterisk-Users] Are there any graphic designers on this list? (B (B (B Hi (B (B I had asked for some help with the Asterisk Assistants (B (B (Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX (B (B and many have offered assistance with translations which I (B am grateful for and like to say thank you again. (B (B However, there hasn't been a single response from a (B graphic designer to offer help with a custom icon. Are (B there any graphic designers on this list at all? If so, (B please take a look at the Wiki above and see if you can (B help. (B (B thanks (B rgds (B benjk (B (B -- (B Sunrise Telephone Systems Ltd (B 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B __ (B GANBARE! NIPPON! (B Yahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (B http://mail.ganbare-nippon.yahoo.co.jp/ (B (B ___ (B Asterisk-Users mailing list (B [EMAIL PROTECTED] (B http://lists.digium.com/mailman/listinfo/asterisk-users (B To UNSUBSCRIBE or update options visit: (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ?
Thank you for repeating this 3 times, I didn't get it the first 2. Please stop spamming the list!!! - Original Message - From: Jean-François Rousseau [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, August 27, 2004 11:03 AM Subject: RE: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ? Hi, I've read the FAQ and tried to find a timing source. So far, I've compiled ztdummy and loaded it sucessfully. But it still not working. All I have is the beginning of the fax. I've tried (HP FAX) -- PSTN -- X100P -- Asterisk -- SPANDSP And (HP FAX) -- IAXy -- Asterisk -- SPANDSP Both do the same error... About a quarter of the page is ok then garbage. The sending machine say that the fax was sent ok. Here is some info that might help troubleshot my problem. ___ Jean-François Rousseau Sys-Tech www.sys-tech.net [EMAIL PROTECTED] Tél. 24h (418) 520-0739 Télec. (418) 520-4554 Ligne directe 1-866-274-4870 Bâtisseurs de solutions informatiques et électroniques [EMAIL PROTECTED] 1-877-969-tech Messages de confidentialité : Ce courriel (de même que les fichiers joints) est strictement réservé à l'usage de la personne ou de l'entité à qui il est adressé et peut contenir de l'information privilégiée et confidentielle. Toute divulgation, distribution ou copie de ce courriel est strictement prohibée. Si vous avez reçu ce courriel par erreur, veuillez nous en aviser sur-le-champ, détruire toutes les copies et le supprimer de votre système informatique. If you can not understand this clause, please contact us for further information because it contain a legal notice. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Underwood Envoyé : 26 août 2004 06:12 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is itpossible ? Am I dreaming ? Stopping in mid page is usually a timing problem. See the spandsp FAQ. Regards, Steve Jean-François Rousseau wrote: Hi , does anybody have successfully received a full fax with spandsp ? I keep having only about a quarter of the page and then the other part is garbage. Does anybody have any solution for this ? Right now I've tried: FAX --- IAXy --- ASTERISK --- SPANDSP And FAX --- PSTN --- X100P -- ASTERISK --- SPANDSP And both don't work, they give me only part of the page BTW, I also tried the fax on a local lan over an IAXy or on the PSTN with an X100P. Is there something I should know about faxing and theses two interfaces ? I also tried to Fax thru asterisk and it didn't work either FAX IAXy --- ASTERISK --- X100P --- PSTN --- FAX Finally my last test: FAX -- IAXy -- ASTERISK -- SIP (Iconnecthere) -- PSTN -- FAX didn't work too. Is there something I should know about faxing and Asterisk ? Should I use a Sipura SIP FXS ? P.S. I did start the ntp server to make sure timing was ok. Thanks in advance ___ Jean-François Rousseau Sys-Tech www.sys-tech.net [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
- Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 5:12 AM Subject: [Asterisk-Users] Overhead Paging All, I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a Paging Unit to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. -Brian The Valcom units don't mention SIP... they must be some proprietary protocol, also says it's Windows based... eww... Anything that you can do to an FXS port on a regular PBX you should be able to do to the FXS port of an IAD (within reason of course). You don't usually connect the speakers directly to the FXS, usually there's a device like an inline power supply that provides voltage and audio to the speakers or bullhorn... The GrandStream HT486 has a nice intercom function which sends a BEEP! before answering. The SPA-2000 also has an intercom but doesn't provide the beep... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
- Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Chris Shaw [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 2:39 PM Subject: Re: [Asterisk-Users] Overhead Paging Chris, What you're talking about is exactly what I'm looking for. I'm interested in the middleware that would sit between the speakers and the IAD. I have found the Bogen TAMB device -- however I was wondering if anyone had any experience with this box. How many speakers can you power off of the unit without needing an external amp, etc... -Brian lol those would be electrical questions... Not sure how to answer those, it would depend on power output of the TAMB device and how much current draw and wattage the speakers require... I know this would work though, as long as the receiver/amplifier has an FXO interface and not an RCA jack like you would find on a stereo system or CD player... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Overhead Paging
- Original Message - From: Brian Pavane [EMAIL PROTECTED] To: Chris Shaw [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 2:39 PM Subject: Re: [Asterisk-Users] Overhead Paging Chris, What you're talking about is exactly what I'm looking for. I'm interested in the middleware that would sit between the speakers and the IAD. I have found the Bogen TAMB device -- however I was wondering if anyone had any experience with this box. How many speakers can you power off of the unit without needing an external amp, etc... -Brian Also I had mentioned earlier that the SPA-2000 and HTX86 IADs have an auto-answer, that's not true, so the TAMB device must recognize ringing voltage and automatically answer... If it can connect directly to a PABX system it most likely does... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
- Original Message - From: Mike Meyer [EMAIL PROTECTED] To: Asterisk Users Group [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:41 PM Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-December/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Mike, I am in no way a SIP expert so please take my comments with a grain of salt, but this would require something attached to the INVITE message sent to the phone. I know that * (with a little patching) can transmit intercom=yes to a SNOM phone indicating that the incoming session wants to use the intercom function... If phones have something similar to this like ringcadence=3 then it could be passed on... I would imagine that the value you pass to the phone is specific to the make and model of phone however... Not all phones would use ringcadence= and not all models would support '3' for example... There has been some discussion of this in the bug lists and here on list, but I'm not sure as to the status of this yet in *... I know for the Cisco phones, there is a variable called ALERT_INFO you can set to change the ring cadence but that's the only phone I know of so far... It would be s cool if Digium made their own brand of phones, or at least commissioned someone like Sayson to do it for them... then we could get a completely *-compatible solution with all of the bells and whistles we want... If we want to go with our own phones, then we would have to sacrifice the fancy features but at least it would still work! Maybe they already have something like this in mind? I would be totally behind it! Especially if it spoke IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Chris Shaw' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 4:42 PM Subject: RE: [Asterisk-Users] Distinctive Ring Cadences This is what you're looking for (Sipura SPA-2000): exten = 201,1,SetVar(ALERT_INFO=bellcore-r1) exten = 201,2,Dial(SIP/201,40) exten = 301,1,SetVar(ALERT_INFO=bellcore-r4) exten = 301,2,Dial(SIP/201,40) This dials SIP-ext 201 with a different ring-cadence when you dial 301. Sipura Supports bellcore-r1..r8, but even those names (and the cadences) are configurable in config. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Distinctive Ring Cadences - Original Message - From: Mike Meyer [EMAIL PROTECTED] To: Asterisk Users Group [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:41 PM Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-Decembe r/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Mike, I am in no way a SIP expert so please take my comments with a grain of salt, but this would require something attached to the INVITE message sent to the phone. I know that * (with a little patching) can transmit intercom=yes to a SNOM phone indicating that the incoming session wants to use the intercom function... If phones have something similar to this like ringcadence=3 then it could be passed on... I would imagine that the value you pass to the phone is specific to the make and model of phone however... Not all phones would use ringcadence= and not all models would support '3' for example... There has been some discussion of this in the bug lists and here on list, but I'm not sure as to the status of this yet in *... I know for the Cisco phones, there is a variable called ALERT_INFO you can set to change the ring cadence but that's the only phone I know of so far... It would be s cool if Digium made their own brand of phones, or at least commissioned someone like Sayson to do it for them... then we could get a completely *-compatible solution with all of the bells and whistles we want... If we want to go with our own phones, then we would have to sacrifice the fancy features but at least it would still work! Maybe they already have something like this in mind? I would be totally behind it! Especially if it spoke IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
Lol all of these would look pretty funny plastered inside a wall... I think you would be better using an ATA adapter and a regular analog DoorPhone or Intercom. Then you get the best of both worlds... it's cheap... and it uses SIP... The GrandStream HT486 and also the 286 has a feature where it dials a specific extension when it goes off hook... this would be PERFECT for an intercom... Implement some dialplan magic and voila! -Chris - Original Message - From: listas iPfone [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, August 23, 2004 9:23 AM Subject: Re: [Asterisk-Users] SIP unphones Polycon SoundPoint IP3000, but it´s h.323 - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 11:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: SIP unphones
Why didn't I think of this before! Better yet, use an analog doorphone or intercom and an IAXy! I haven't had the pleasure of using one yet but I'll bet they can do some pretty neat tricks, especially since they're speaking IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: newb question regarding DTMF
- Original Message - From: Erik Anderson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 9:03 AM Subject: [Asterisk-Users] Re: newb question regarding DTMF On Mon, 23 Aug 2004 18:20:58 -0500, Erik Anderson [EMAIL PROTECTED] wrote: Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? Bump Any advice? Is the dtmfmode= line in your sip.conf the same as the dtmfmode in your X-Lite Client? If not, * will not be able to understand DTMF coming from the client once an extension has been dialed and pushed onto the SIP stack... I believe X-Lite defaults to using RFC2833 DTMF, so make sure in your X-Lite context in sip.conf that you have set dtmfmode = rfc2833... Be sure to check out the samples provided in the Asterisk/configs directory as well as... http://www.voip-info.org/wiki-Asterisk+config+sip.conf -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
Check out my ATA idea though, with a regular cheap analog doorphone and a HTX86 or even Sipura, you can program the ATA to dial an extension as soon as the button on the intercom is pressed and then with some extension logic you can do neat things... You can get a doorphone anywhere even radio shack I think and the HTX86 is like $60-70... - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:16 PM Subject: RE: [Asterisk-Users] SIP unphones Thank you -- funny thing is, I had the same bookmarked, but it just seemed too expensive for the application -- for $300, I can stick a cheap IP phone in a hole in the wall :) I think it's time to get a Budgetone. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Monday, August 23, 2004 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP unphones I recently saw something just like this and I had it bookmarked... It looks like what you're talking about, but I don't think it uses SIP. Rather some proprietary protocol that transmit RTP... I could be wrong... Check it out... http://www.digitalacoustics.com/lanplay.htm I would agree that it really should be SIP, you wouldn't want to have to rip it out of the wall when the protocol becomes obsolete or when a SIP-Compliant alternative comes out... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MailMan slowness...
Ok what the flaming hell is up with the MailMan? It's taking over a day to send posts now, before it was at most a couple hours... Is digium doing maintenance on that server or something?? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Adit 600
Are you talking about an ETHERNET crossover cable? You can't use an ethernet crossover, the pinouts are different! Look here for a wiring diagram: http://www.gcom.com/home/support/t1crossover.html -Chris - Original Message - From: Craig Neumanns To: [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 2:31 PM Subject: [Asterisk-Users] Asterisk with Adit 600 Hello, I am connecting Asterisk to an Adit 600 via a T100P. Unfortunately I am not able to get any lights on the T100P and the Adit 600 only showsred. I have already modprobed the t100p. Has anyone successfully connected these before? I've tried both a cross-over cable and a standard cat 5 cable, but I can't seem to get them to work. Any suggestions? Thanks! Craig Do you Yahoo!?New and Improved Yahoo! Mail - 100MB free storage! ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711u-law. Use RFC2833
Is this message coming from * or from Kphone? That's the stupidest error I've ever seen, of course In-Band DTMF works over U-Law and A-Law... in fact it ONLY works on U-Law and A-Law and not reliably on any other... If this error is coming from *, and I can't believe it is, then you have a codec problem... If the error is coming from Kphone you will need to complain to them that their DTMF generation/handling code is FUBAR... I use In-Band DTMF on my GrandStreams all the time and it works perfectly... Even in Comedian Mail... -Chris - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 2:30 PM Subject: Re: [Asterisk-Users] Inband DTMF is not supported on codec G.711u-law. Use RFC2833 On Tue, 2004-08-24 at 14:00, Eric Wieling wrote: On Tue, 2004-08-24 at 12:03, Steven Critchfield wrote: Inband DTMF is not supported, Use RFC2833 Go search your kphone configs and fic it to use some out of band signalling of DTMF. kphone does not support RFC2833 DTMF, only inband DTMF. Then don't use kphone, it will only cause you more grief down the road. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remotely change call forward
- Original Message - From: Russell Horn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 2:55 PM Subject: [Asterisk-Users] Remotely change call forward Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R. I see 2 ways of doing this... 1 would be an AGI script that uses MySQL or PostgreSQL to store and retrieve forwarded numbers.. The 2nd and to me easier way would be to use AstDB and some extension coolness to store the forwarding number, and then implement some kind of vertical service code that you can dial when you call, someting like *78EXTENSIONNXXNXX, then feed the NXXNXX part of the extension to DbPut(features/fwd-EXTENSION) or something like that... Check these out for some ideas/hints gimme a holler if you need more help! http://www.voip-info.org/wiki-Asterisk+cmd+DbGet http://www.voip-info.org/wiki-Asterisk+cmd+DbPut http://www.voip-info.org/wiki-Asterisk+cmd+AGI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Vonage
I hold no ill will towards Vonage but I have to say honestly... ewww... They've already made their feelings quite clear by refusing to allow people to bring their own devices and taking steps to even hide their SIP servers (changing the port from the RFC standard 5060 to 5061 for example.) Why not go with someone who's actually willing to allow you to use Asterisk and any phone you want like NuFone, BroadVoice, IconnectHere or a host of others instead of trying to hack Vonage... Again... Ewww... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 4:06 PM Subject: RE: [Asterisk-Users] Asterisk to Vonage Yes, search google for asterisk vonage working site:lists.digium.com -Original Message- From: Paterson, Mark [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 11:19 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk to Vonage I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with the vonage sip proxy. Any examples would be greatly appreciated. Rgs, mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
Is this for a remodel or a new home? Depending on your home's telephone layout, some homes have a 66-Block or a 110-Block in the basement or crawlspace under the stairs... you could make your setup look really pretty by wiring the intercom into the 66-Block and then wiring the other end of the 66-block to a wall jack in the room where you keep * box, then you can plug the sipura into that... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Chris Shaw' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 4:08 PM Subject: RE: [Asterisk-Users] SIP unphones Yep, great idea, that's what's next -- and I have two extra extensions (Sipura) -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP unphones Check out my ATA idea though, with a regular cheap analog doorphone and a HTX86 or even Sipura, you can program the ATA to dial an extension as soon as the button on the intercom is pressed and then with some extension logic you can do neat things... You can get a doorphone anywhere even radio shack I think and the HTX86 is like $60-70... - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:16 PM Subject: RE: [Asterisk-Users] SIP unphones Thank you -- funny thing is, I had the same bookmarked, but it just seemed too expensive for the application -- for $300, I can stick a cheap IP phone in a hole in the wall :) I think it's time to get a Budgetone. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Monday, August 23, 2004 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP unphones I recently saw something just like this and I had it bookmarked... It looks like what you're talking about, but I don't think it uses SIP. Rather some proprietary protocol that transmit RTP... I could be wrong... Check it out... http://www.digitalacoustics.com/lanplay.htm I would agree that it really should be SIP, you wouldn't want to have to rip it out of the wall when the protocol becomes obsolete or when a SIP-Compliant alternative comes out... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
Re: [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet
Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully place calls using voipjet with everything except the grandstream. When I place a voipjet call with the grandstream, I can hear the party I'm calling, but they can't hear me. I have tried all the different codecs the grandstream supports without luck. I am running the 1.0.5.10 firmware. I've emailed voipjet support about it, but they don't have one. Can you send your * debug output? It's kinda hard to figure out what's going on without it... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using ChanIsAvail
Title: using ChanIsAvail
Looks correct to me, I'm using a similar setup...
Sounds like maybe it's a bug in the ChanIsAvailApp, like maybe it's
hardcoded to look in sip.conf...
-Chris
- Original Message -
From:
Poul Pedersen
To: [EMAIL PROTECTED]
Sent: Monday, August 23, 2004 3:15
AM
Subject: [Asterisk-Users] using
ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular
extension is available in the sip channel
I am using MySQL to hold my SIP friends. and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip
channel, I will send the call somewhere else
my extensions file contains the following:
exten = _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten = _[123]XX,2,dial(sip/${EXTEN},30) exten =
_[123]XX,102,Dial(IAX2/sip01-xx:[EMAIL PROTECTED]/${EXTEN})
if I understand ChanIsAvail correctly this should give med
following:
if i dial extension 111, and that is a local extension, it
dials the sip channel on the other hand, if extension
111 is not avaliable in the local sip channel, it dials on IAX2
But it does not work, if 111 is not a local extension the dial
in priority 2 returns with -1, in my opinion it should never have been
executed
when I have all SIP frinds in sip.conf it works, but it does
not when using MySQL
is this a bug, or is ChanIsAvail not intended to work when SIP
frinds are in MySQL ??
Kind regards
Poul Pedersen
___Asterisk-Users
mailing
list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo
UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 22, 2004 8:12 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Sat, 2004-08-21 at 16:37, Paul Concepcion wrote: well, that's our setup (8 analog lines - channel bank - t100P), so it looks like DNIS is out of the question. We do have 8 phone numbers though. Could we have a 1-800 number direct to each of those, then do what you suggested with contexts? What would happen if two people dialed 1-800-a if 1-800-a was pointed to just one phone number? Depends on hunt groups and such. If you have rollover/hunt groups, pointing a 1800 to a number is not very useful for getting DID or DNIS functionality. The different context solution was based on the idea of making each incoming analog line have it's own logical seperation in the dialplan. The trouble is, as you roll from one busy line to the next, there is no information about what group the person dialed into. If you where to split your hunt group into 2 - 4 line groups without talking to the telco, you could fill group 1 up and then be rolling into group b. Same works the other way with wrap around hunting. If you don't have hunt group functionality, and you point a 1800 number to a analog line, then the second phone call will hit a busy signal. I'm using a similar setup here, we have 3 companies in this building. We're using a Merlin Legend PBX with FXO modules. Our incoming lines come from a T1 which terminates on an ADIT 600. It is then split into lines through FXS cards in the ADIT... Company A has 5 lines, the first of which has the 1-800 number pointed to it. It is set up on a linear hunt group to the other 4 lines. No matter what line the call comes in on, since it's in that first set of 5 lines, the PBX answers with Company 'A' IVR... * can do the same thing, I would group the first 5 channels into 'g1' for example, then place them in a context like [companyA]... Company B has 3 lines, same thing only set up on a separate linear hunt group so that it doesn't roll into the first 5 lines or the next 8 lines... Company C has 8 lines... you get the idea... I'm not sure how many companies you have or how many 1-800 numbers you're using... Obviously this is not the ideal setup because it requires the different companies to have a fixed amount of lines whether they use them all or not... A better solution would be a PRI with DNIS but this is what we have to work with and it seems to work well... Like Steven said if you don't have hunt groups, then when someone calls a number and another person calls that same number, the 2nd person will get a busy signal... At least with the way our hunt groups work, the hunt will keep looking in a linear fashion until a line becomes free (resulting in the person hearing ringing)... Hope this helps! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
I recently saw something just like this and I had it bookmarked... It looks like what you're talking about, but I don't think it uses SIP. Rather some proprietary protocol that transmit RTP... I could be wrong... Check it out... http://www.digitalacoustics.com/lanplay.htm I would agree that it really should be SIP, you wouldn't want to have to rip it out of the wall when the protocol becomes obsolete or when a SIP-Compliant alternative comes out... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
...Today there's no valid reason to use telnet over ssh. Was there ever a valid reason? Maybe export restrictions on crypto? I've never EVER used telnet or rlogin, SSH is so much nicer anyway... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
LOL it was so long ago, I didn't think about that reason... :) - Original Message - From: Walt Reed [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 9:13 AM Subject: Re: [Asterisk-Users] telnet and Root On Fri, Aug 20, 2004 at 08:40:43AM -0700, Chris Shaw said: ...Today there's no valid reason to use telnet over ssh. Was there ever a valid reason? Maybe export restrictions on crypto? I've never EVER used telnet or rlogin, SSH is so much nicer anyway... Yeah. Some of us were around before ssh existed. :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
- Original Message -
From: James Freire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 9:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi All,
I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.
Thanks,
James
Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...
What I envision is doing something like this...
[verticalservice]
exten = *78,1,DbGet(${dnd}=features/dnd)
exten = *78,2,DbPut(features/dnd=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
exten = *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh = *78,103,DbPut(features/dnd=1)
exten = *78,104,Playback(pbx-dndenabled)
exten = *78,105,Hangup()
exten = *79 ... etc...
Then in your extension calling macro, you're going to want to check against
the DB like this...
[macro-insidedial]
exten = s,1,DbGet(${dnd}=features/dnd)
exten = s,2,DbGet(${fw}=features/fw)
exten = s,3,Dial(${ARG1},25,tT)
exten = s,4,VoiceMail(u${ARG1})
exten = s,5,Hangup()
exten = s,102,GotoIf($[${dnd} = '1']?200:2)
exten = s,103,GotoIf($[${fw} = '1']?300:3)
exten = s,104,VoiceMail(b${ARG1})
exten = s,200,VoiceMail(b${ARG1})
exten = s,201,Hangup()
exten = s,300,Dial(SIP/[EMAIL PROTECTED],60)
exten = s,301,Congestion()
be sure to include [verticalservice] in your inside-office context...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Wouldn't you need to track each extension? something like:
exten = *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten = *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
etc.?
Yep! good catch! that's why I asked someone to correct me, I was in a hurry
and this was an on-the-fly kind of example...
You would need to do something like this, or make a key like
features/dnd-${CALLERIDNUM} would be best... Would also work for
forwarding...
-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
If you really want to be able to telnet in as root, locate telnetd.conf or somesuch and it should be in there somewhere as a yes/no. (It is for ssh anyway..) No, not under any distro I'm familiar with... It's under /etc/securetty... You add the tty of the device you want to allow root access to, like pts/0... DON'T DO THIS THOUGH, unless you don't care that your root password will be sent PLAINTEXT over the internet... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service codes are mentioned anywhere... A quick look through chan_zap reveals all of them... So for right now it's not implemented in SIP... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
Why not use separate contexts for these lines in zapata.conf? Seems way simpler to me... http://www.voip-info.org/wiki-Asterisk+config+zapata.conf -Chris - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 12:33 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote: Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on who called 1-800-777- and who dialed 1-800-555-, for example. Yes, but it depends on what type of telephony signalling you are using as to whether or not you can get that information. Tells us about your PSTN connection. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
True, very true... If it's PRI then you will get DNIS/DNID from the D-Channel... If they're doing anything other than PRI though, like a regular T1 into a channel bank (or into a TE100P or TE40xP) , this would work... Lines would be assigned to a specific channel and they could be separated out with contexts... Not as pretty as using DNIS but it would work... -Chris - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:29 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 15:23, Chris Shaw wrote: Why not use separate contexts for these lines in zapata.conf? Seems way simpler to me... http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Who said they are seperate lines. 1-800 numbers can just be redirects to other lines. In that case you have to have a different signaling method to determine 1800-a on line b is different from 1800-c on line b. In a PRI circuit, the lines are just channels, all signaling data is run over the D channel and any line can be any number routed to that circuit. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 12:33 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote: Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on who called 1-800-777- and who dialed 1-800-555-, for example. Yes, but it depends on what type of telephony signalling you are using as to whether or not you can get that information. Tells us about your PSTN connection. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:37 PM Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Chris Shaw wrote: I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service codes are mentioned anywhere... A quick look through chan_zap reveals all of them... So for right now it's not implemented in SIP... Well, here we stumble over the SIP religion again. First, a phone connected to an RJ11 jack in a Digium card is a stupid phone. All the intelligence lies in the zaptel driver and asterisk. Most SIP phones are more clever (at least expected to be much more clever than the GS :-). Look at the SIPURA, where you are able to implement vertical service codes in the SIPura. Asterisk should not bother with DND and forwards, the SIP phone does. Just send the call to the phone. Some of these phones are complete Linux systems with IPsec, multiple lines and a lot of routing intelligence. There's also a discussion between Asterisk developers on whether these codes should be fixed in the channel or in the dial plan. At least, they should be configurable since there's no global standard (again). Or there may be, but there are still differences between countries and providers... * Executive summary: SIP is designed for very intelligent end-points. * A PBX with analogue lines is designed for central intelligence. * Asterisk will always be in the middle of these kind of discussions, and it'll be fun each time we try to sort it out. /Olle No, I agree completely with the way it works now, in fact I think it SHOULDN'T be implemented in SIP myself... Doing it in the dialplan (if your phone doesn't support it) works fine and doesn't break anything (that's the key right there). We need some more docs on how to do different things and I'm sure many people could contribute those, myself included... Some already have... The only thing is, if any of the apps you've written in your dialplan become obsoleted or change syntax, your whole implementation will get screwed over... I guess that's true with anything though... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
Except for what you said about 1-800 numbers pointing to the same line... nevermind I'll shut up now... -Chris - Original Message - From: Chris Shaw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:47 PM Subject: Re: [Asterisk-Users] determining what number was dialed? True, very true... If it's PRI then you will get DNIS/DNID from the D-Channel... If they're doing anything other than PRI though, like a regular T1 into a channel bank (or into a TE100P or TE40xP) , this would work... Lines would be assigned to a specific channel and they could be separated out with contexts... Not as pretty as using DNIS but it would work... -Chris - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:29 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 15:23, Chris Shaw wrote: Why not use separate contexts for these lines in zapata.conf? Seems way simpler to me... http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Who said they are seperate lines. 1-800 numbers can just be redirects to other lines. In that case you have to have a different signaling method to determine 1800-a on line b is different from 1800-c on line b. In a PRI circuit, the lines are just channels, all signaling data is run over the D channel and any line can be any number routed to that circuit. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 12:33 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote: Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on who called 1-800-777- and who dialed 1-800-555-, for example. Yes, but it depends on what type of telephony signalling you are using as to whether or not you can get that information. Tells us about your PSTN connection. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura endpoints
My first and only unit cooked itself. It literally melted the casing. Sipura replaced it very promptly though. Wow that's bad! Bad power supply? Hmmm I don't think he meant 'flame' literally... lol... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
http://www.voip-info.org/wiki-asterisk+pbx+functions http://www.voip-info.org/wiki-asterisk+vertical+service+activation+codes - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 3:02 PM Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi, sorry for interruption, but are there any guides for all possible Asterisk PBX functions that are available with no particular dialplan handling ? Thanks, Robert. - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 6:09 PM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding macros causes ringing to fail
Ok.. this is really wierd... I just cleaned up my dialplan a bit by adding some macros with a strange side effect... On my incoming context which has no macros in it, far end ringing used to work... now that I have macros defined, far end ringing has stopped working all together... The macros DO work, but when they transfer, the far end ringing sounds terrible and even skips a few rings... If I remove the macros and put things back the way they were, things return to normal... This is VERY wierd... -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding macros causes ringing to fail
I forgot to mention, these are SIP calls, I use a Pure SIP configuration... it's so strange, if I remove macros and do the EXACT same thing without macros, it works perfectly! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work?
Title: Does Granstream BT100 Conference Button Work? Nope, it does nothing... It's not an * problem either, the button just does nothing... I think they're planning on making it work in a future release, don't quote me on that... for now it just occupies space.. -Chris - Original Message - From: James Freire To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 12:53 PM Subject: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James
Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work?
Title: Does Granstream BT100 Conference Button Work? I'm sure you could, you could also use a MeetMe conference room... - Original Message - From: James Freire To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 1:34 PM Subject: RE: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Could I use the Flash button to do conferencing then??? If so.. how? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Chris ShawSent: Thursday, August 19, 2004 4:28 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Nope, it does nothing... It's not an * problem either, the button just does nothing... I think they're planning on making it work in a future release, don't quote me on that... for now it just occupies space.. -Chris - Original Message - From: James Freire To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 12:53 PM Subject: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James
Re: [Asterisk-Users] Does Granstream BT100 Conference Button Work?
Well it does. It hangs up the connection, on my phone. Latest firmware. : ) - -- Steve lol... YAY!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup?
OK... Don't know what happened there, but I can blame it on OE's LAMENESS... Anyway, as I was saying I have 3 BT100s and none of them do that, it must be a firmware issue. Maybe it's a wierd isue with auto answer? -Chris - Original Message - From: Chris Shaw [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 2:32 PM Subject: Re: [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup? Does Granstream BT100 Conference Button Work?Sounds like a firmware thing to me, I have 3 of them and none of them do that.. - Original Message - From: Kanuri, Seshu To: [EMAIL PROTECTED] Sent: Thursday, August 19, 2004 2:29 PM Subject: [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup? Hi Folks! I have another problem with BT100 Phone. Whenever someone calls me, it rings once and stops. But the call is still on hold till I pickup. How do I increase the number of rings? Is this a * problem? or BT100 Issue? Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Freire Sent: Thursday, August 19, 2004 3:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Does Granstream BT100 Conference Button Work? Hi All, I have tried searching everywhere but I cannot find a definitive answer as to if and how the conference button works on the BT100. Could anyone be kind enough to fill me in on some info on how to use the conferencing feature, as well as any configuration in asterisk thats needed, on this phone? Thank you, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inband announcement of parking slot from app_parkandannounce?
Be aware that if you want to use SIP (you didn't mention you were) Park
still doesn't play nice with SIP transfers... It works, but you never hear
the announced parking slot... I think this is being addressed though... Also
there's BKW's nice valet_parking.so application which has more features than
regular parking anyway... Like the ability to specify multiple parking lots,
etc...
-Chris
- Original Message -
From: Kris Boutilier [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 2:32 PM
Subject: RE: [Asterisk-Users] Inband announcement of parking slot from
app_parkandannounce?
Couldn't see the forrest for all the fascinating tree-like applications
that
are out there:
For future reference, see:
http://www.voip-info.org/wiki-Asterisk+call+parking
:-)
-Original Message-
From: Kris Boutilier [mailto:[EMAIL PROTECTED]
Sent: August 11, 2004 1:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Inband announcement of parking slot from
app_parkandannounce?
I'm trying to use Asterisk app_parkandannouce to build a global parking
pool from within a couple of Norstar PBXes.
{clip}
So, the question becomes: How do I structure my extensions.conf to
convince
app_parkandannounce to play it's message on the incoming channel rather
than
using dialback? Have I simply missed a switch somewhere, or will I have to
use some sort of meetme bridge configuration to short-circut the incoming
call with the announcement?
{clip}
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GrandStream BT101 Attended Transfers
I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?
I believe you can do that kind of thing with SIP, using the
SetVar(${SIP_CODEC}=CODEC}) in your extensions.conf... So, For example, if
the extension of your card machine were say '100' then you would do
something like this
[outgoing]
exten = _1NXXNXX,1,Gotoif($[${CALLERID_NUM} = '100']?2:3)
exten = _1NXXNXX,2,SetVar(${SIP_CODEC}=ulaw)
exten =
_1NXXNXX,3,Dial(IAX2/user:[EMAIL PROTECTED]/$EXTEN,30)
I don't know if this works with IAX...
-Chris
- Original Message -
From: Kris Boutilier [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 19, 2004 3:11 PM
Subject: [Asterisk-Users] How to run different codecs between the same
endpoints on an IAX trunk?
Or perhaps how to configure and refer to two parallel IAX trunks with
different codecs?
I have a situation where I'm using G.729A as my IAX trunking codec. Now I
need to push some short duration, low bitrate modem traffic over the link
(a
credit card terminal). Obviously the modem audio isn't going to survive
the
G.729 codec process intact, so for the times the device is used I'd like
to
service calls from that device (and only that device) with a higher-data
rate codec.
Any suggestions?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More on Broadvox
I have to ask... why are you trying to get a SIP provider to work if they clearly aren't interested in supporting Asterisk? -A. Most likely because they don't want to loose their DID number... A very valid reason! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream BT101 Attended Transfers
This is lame, but AFAIK, the only way to do it is: - Press Flash button dial party to transfer to - inform party about call ask that party to hangup - press flash again to return to original caller - press transfer dial number to transfer to - press send Ryan Nevermind, they say it uses 2 kinds of blind transfers. REFER and BYE/ALSO... Couldn't you after talking to the 3rd party hang up on them so that * disconnects the channel and THEN press FLASH to return to the first call, then transfer? Is attended transfer possible with the 'T' dial function or is it just blind? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] residential sip phone
If you don't want to have to talk through your computer (ala X-Lite/Pro) then there's the GrandStream BT101, it has some minor quirks (e.g. 3-way calling does not work even though there's a button) but they should be fixed in later firmware releases. Also GrandStream makes ATA devices (devices that make your regular analog or wireless phone speak SIP). These phones can be found for as low as $65 USD and have EXCELLENT voice quality and can even work behind NAT... There's also SIPURA for the ATAs IPDialog SipTone II is another SIP Phone... Uniden makes one also, both of these are over $100 apiece... - Original Message - From: John Williams To: Asterisk Users List Sent: Thursday, August 19, 2004 3:10 PM Subject: [Asterisk-Users] residential sip phone Dear List, Can anyone recommend a sip phone for residential use? (asterisk home pbx) Thanks!!!
Re: [Asterisk-Users] GrandStream BT101 Attended Transfers
BTW, Ryan, Thanks for the info! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream BT101 Attended Transfers
Sure, so long as that person gets the hint to hang-up when they hear the congestion tone... I see what you mean... You have to be careful too, I've had a GrandStream drop a channel (I'm assuming without sending * a BYE) and then * will keep that channel open and there's no way (short of issuing a soft hangup) to hang it up... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New $89 VOIP phone
- Original Message - From: Stefaan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 11:23 PM Subject: Re: [Asterisk-Users] New $89 VOIP phone From: Andrew Kohlsmith [EMAIL PROTECTED] Either way a decision needs to be made. There's no magic fairy gonna come down and wiggle her pretty lil' ass over the walls and you magically have dual Cat5e to every desk and some great POE-injecting switches upstairs. :-) Those fairy's do actually exist, well, kind off that is ;-) Do you know those ethernet cable splitters? Is splits 1 ethernet cable into 2 ethernet connections by using all 4 pairs of the cable. Put one at your desk, and one at your switch, et voila; 2 independent ethernet connections over one cable. You could also do this without those splitters by splitting 2 pairs of wires to 2 connectors on both side of the cable. You're kidding right? There's a reason why category 5 cable is twisted the way it is... to eliminate or greatly reduce RF crosstalk on the wires... Now what would happen if you split those wires between 2 different signals and kept them tightly packed together? Can we say random data corruption, mysterious errors and terrible performance? Bingo! Again, this kind of thing might be ok for a small home network, but you can't seriously suggest it for a business... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] What's changing /etc/hosts?
Why not try 'lsof' to see what processes might have it open or might be writing to it... - Original Message - From: Mark Woods [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 18, 2004 10:09 AM Subject: Re: [Asterisk-Users] [OT] What's changing /etc/hosts? do a 'ps -ef | more' or 'ps -aux | more' and look at the processes that are listed to see if there is something running that might be doing it. Otherwise, I'd approach it by going through each of the startup scripts (rc#.d, etc.) and then each application's startup scripts. A bit tedious, but... -Mark Occasionally my /etc/hosts file gets corrupted. The IP address and the host name switch positions with the host name to the left. What this happens, my 7940 phones won't register. Fixing /etc/hosts allows the phones to register. Do any of you Linux gurus know who is corrupting the hosts file? Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: New $85 VOIP Phone
Back to the ACTUAL TOPIC of this thread... This phone looks kinda nice, where can one get hold of it? How about it's * compatibility? I realize that it says it does things like 3-way conference and attended transfers, but how about in *? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
