Re: [Asterisk-Users] Store caller IP in CDR
Barry Fawthrop wrote: From: Olle E. Johansson [EMAIL PROTECTED] snip Check the CDRuserfield - it's a free field in the CDR you set in the dialplan or from a script. How would you set the CDRuserfield from the dialplan exten = ? http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd%20setcdruserfield /O ___ 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] Snom 200
Barry Fawthrop wrote: For some reason MWI, wants to dial [EMAIL PROTECTED], I have not exten or account asterisk ???, can't even find where this is set ? http://www.voip-info.org/wiki-Asterisk+phone+snom /O ___ 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 T1 problem (Avaya - asterisk IVR)
James, Thanks so much for taking the taking the time to help me figure this out and learn something. It is a Definity, but I'm not sure about the card -- just a basic t1 card is all I know (on Monday I could get more info). Is there a command to find out which card is installed? or if that is enough to get started with the commands? Thanks again, Jeb On Mar 20, 2004, at 11:37 PM, James Coberly wrote: What Avaya card are you using? What model of system? Definity, Merlin, etc? With this I should be able to send you the base commands to review the card slot settings for the PXB 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] Echo Cancellation (Newbie Qu)
Hi, I've tried all the other methods of dispelling the x100p echo mystery such as echo training rx tx gain through ztmon and swapping POTS lines etc etc. Can someonemail me a step by step guide to changing the echo cancellation algorithms such as Mark,Mark2, Steve etc. Muchos! Taff. Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now
Re: Subject: Re: [Asterisk-Users] firefly softphone
On Sun, 2004-03-21 at 04:29, Chris Jones wrote: In my opinion just dump firefly and use something that works. I did. Works for me, receives calls makes calls, doesn't make the coffee. HP Omnibook, W2KPro via Wifi using IAX2 -- Dave Cotton [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] T100P T1 problem (Avaya - asterisk IVR)
Hi Jeb, Have a look on: http://www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya I think it's what you need. Marc -- Message original -- To: [EMAIL PROTECTED] From: Jeb Campbell [EMAIL PROTECTED] Subject: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR) Reply-To: [EMAIL PROTECTED] Date: Sat, 20 Mar 2004 19:36:26 -0500 Hello all, I'm having a problem with a T1 connection to a Avaya PBX (asterisk is an IVR). I could not get pri working and now I'm simply trying to get asterisk working with fxs_ks. Questions: 1. Is there anyway to troubleshoot or see what is being sent on the T1. zttool shows no errors, and the Avaya rings the t1 if wct1xxp is loaded, and gives busy if it is not loaded -- so I think something is being sent down the line, I just don't know what it is (asterisk -vc shows nothing) 2. Is there anyone with Avaya PBX - asterisk experience on the list? I'm remote, but I can login -- I just don't know what commands can troubleshoot the connection. Config: zaptel.conf (relevant section) span=1,1,0,esf,b8zs fxsks=1-24 zapata.conf [channels] context = demo switchtype = national signalling = fxs_ks group = 1 channel = 1-24 Thanks for any time and help Jeb Campbell [EMAIL PROTECTED] Cell: 865-385-1437 ___ 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 i do voice chat without using the hardware
Hi, Thanks a lot for your help. After installing iaxComm, When I test Asterisk typing # asterisk c I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory ) Why i am getting this error? How can i tackle this error? Before installing the iaxComm, i will get the CLI prompt. Now it's not getting it. So please help me to solve this problem. Thanks Regards, Sur --- Michael Van Donselaar [EMAIL PROTECTED] wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman [EMAIL PROTECTED] wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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 i do voice chat without using the hardware
Hi Girish, Thanks a lot for your help. I am also made an attempt to install Linphone, but i got an error. According to your suggestion, i installed softphone from zultys.com. That's fine. I had gone through the http://www.automated.it/guidetoasterisk.htm; link and got more information from this link. Now i am facing the problem is that When I test Asterisk typing # asterisk c i am getting a display as Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. When i type asterisk -r, i am NOT getting any CLI prompt now ... getting display as Asterisk CVS-03/18/04-18:01:45, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = So what should i do now? I don't have any information to debug this. I am new to this area, so please give me a help to solve this problem. I got an advice from the another expert, saying that i have to install iaxComm (uses asterisk's native IAX protocol). So I tried to install that, it's also creating some problem (Saying that Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) ). I am waiting for your reply. Thanks Regards, Suresh --- Girish Gopinath [EMAIL PROTECTED] wrote: Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can i do voice chat without using the hardware Date: Fri, 19 Mar 2004 05:50:00 -0800 (PST) Thanks a lot for your valuable information. I will go through it once again. Still i don't have any idea to connect two PC's. Hope i may get help from you. For configuring 2 softphones with Asterisk see this link: http://www.automated.it/guidetoasterisk.htm That helped me a lot in learning Asterisk. It explains configuring your sip phones with Asterisk. Is there any softwares like X-lite for Linux? Yes, I think you can use linophone. But i was not able to install linophone because of some make issues. Also i have tested the softphone from zultys. It works well with Asterisk. You can get it from their web site:http://www.zultys.com Regards, Girish _ Catch the formula fever! Get all the latest news. http://www.msn.co.in/formula2004/ Right here on MSN. ___ 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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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] High latency from Europe, 500-800ms.
You are right to suspect codec issues here. What codec are you using at the various endpoints? Make sure that the Asterisk box is set up with the correct codecs in the conf files, otherwise it will try to transcode and this will often cause bad audio quality like you mentioned. If you're using G729, make sure that you don't have any rTt or other options in dial enabled, otherwise, Asterisk will proxy the media. I've read in previous threads that the jitter buffer is broken in iax and we tried with and without and it was much better without. On Mar 19, 2004, at 8:57 PM, [EMAIL PROTECTED] wrote: Maybe I'm not articulating myself well. The 7940 on the same network in Europe *works great*, no problems, sound is perfect, even with the higher latency. If I take that 7940 and have it connect to a *local* Asterisk server, which connects to the states, it sucks. The 7940 though, connecting directly to the states, works great. Bill Not all sat connections are one way. But the issue with sat connections is *drumroll* latency! As the signal is beeing relayed over the sattelite this will cause latency. Also if the sat service is not providing enough downstream it's bad too. I would definately look into getting your network straighend out first. There are many factors. Is your connection shared? What speeds? Let say it like that if you have people on your local lan using bandwith or running peer 2 peer filesharing stuff this will take away your upstream speed. Do some tests. ___ 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] Packet8
On Sat, 20 Mar 2004, Zac Amsler wrote: I know this issue has been address before, but I can not find someone who has the answer. I am trying to get my * server to authenticate directly to packet8. I was very close to them actually giving me the information and possibly using them for my SIP - PSTN termination, but that fell through. They didn't think they had enough bandwidth. (LOL) There are a few questions that I would like to know answers to. - Does anyone currently have a working implementation in which asterisk authenticates to pakcet8? (Making and receiving calls via packet8) If so, could you please share? Hi, I used to use * with Packet8 - it took some fixes to the * SIP implementation but those are in the CVS long time ago. But then Packet8 started sending emails complaining about my foreign UA software and threatening disconnection. I suppose this was to do with stopping people pumping millions of minutes through one flat-rate account. Ironically, I was on the per-minute rate. Anyway - I disconnected and concluded that Packet8 didn't want to deal with us. No loss to us - providers like Nufone and Magrathea and others are there to take our business. Steve ___ 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] SoftFAX/spandsp
Hi, I have received more excellent problem report information, and I have resolved a number of issues affecting my soft FAX machine when working with various models of real FAX machine. The code now seems to be working with a much greater range of fax machines. A problem affecting the reliability of multi-page fax receive has also been corrected. You can get the latest code from ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1f.tar.gz The application program ftp://ftp.opencall.org/pub/spandsp/app_rxfax.c has also been updated to remove a redundant variable. After building and installing the latest spandsp, rebuild the asterisk applications, as some data structures have changed. Regards, Steve ___ 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] Discriminate on IAXTEL dial-in
Hi, I have two IAXTEL accounts, which I activate with: register = alphanet:[EMAIL PROTECTED] ; 1-700-895-5211 register = cril:[EMAIL PROTECTED] ; 1-700-669-1152 when someone dial this number, it goes through the iaxtel-user context. In extensions.conf, I tried: exten = 17008955211,s,Goto(iaxtel-guest,s,1) exten = 17006691152,s,Goto(isdn-free-dial-out,,s,1) unfortunately it doesn't seem seem to work easily, maybe because IAXTEL doesn't send me the called ID ? ___ 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 i do voice chat without using the hardware
On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks a lot for your help. After installing iaxComm, When I test Asterisk typing # asterisk c Are you running iaxComm on the same machine as asterisk? You can't do that. I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory ) Why i am getting this error? How can i tackle this error? Before installing the iaxComm, i will get the CLI prompt. Now it's not getting it. So please help me to solve this problem. Thanks Regards, Sur --- Michael Van Donselaar [EMAIL PROTECTED] wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman [EMAIL PROTECTED] wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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] If you know your party's extension # please dial it now ...
Hi all,
I've built the usual press one for sales, 2 for support IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the options.
It seems though that one can only press one number before the IVR moves to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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Re: [Asterisk-Users] Discriminate on IAXTEL dial-in
On Sun, 2004-03-21 at 07:27, [EMAIL PROTECTED] wrote: unfortunately it doesn't seem seem to work easily, maybe because IAXTEL doesn't send me the called ID ? You are correct, IAXtel does not send the called number. Calls from both IAXTel accounts will fall into the s extension. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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 T1 problem (Avaya - asterisk IVR)
Jeb, Do you know what slot it is in? Carrier A (top) or B (bottom)? We should take this off list though and reply to me directly from this point, since this is not really * related now. There are 2 ways to do this: At the system propmt type: list configuration ds1 (will list all DS boards in the system) list configuration all will give you all boards in the system. FInd the one related to the slot you are connected to. Or if you have a restricted shell: You can look at the back of the unit, locate the amphenol you connected, there is a no. (slot #) Locate the card on the front of the unit in that slot. Should be marked TNXXX 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
Re: [Asterisk-Users] If you know your party's extension # please dial it now ...
On Sun, Mar 21, 2004 at 08:37:25AM -0500, Mark Phillips wrote:
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
It works fine for me using V1.0-Stable
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
This will wait for tones before timing out. You decide how long
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
If you try it, it should work!
I'm not using a wildcard in my extensions, I include the context that defines the
extensions for the internal phones. ie one of my extensions is 3010. So, the IVR has
an extension 3 to dial a specified group or extension 3010 for a specific extension.
I'm using contexts and I build my incoming context by including various contexts that
are required for the IVR.
Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
You say no one can dial your extensions? Well no one should be able to,
your extensions aren't listed in the IVR.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Phillips
Sent: Sunday, March 21, 2004 8:37 AM
To: Asterisk Users
Subject: [Asterisk-Users] If you know your party's extension
# please dial it now ...
Hi all,
I've built the usual press one for sales, 2 for support IVR
which works fine but I'm having difficulty in allowing
callers to type in whole extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the
IVR below (just in case someone wants one). The welcome
message states callers should type in the extension number
they want or choose from the options.
It seems though that one can only press one number before the
IVR moves to the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I
can't have any menu choices beginning with 3 or 4. Would this
be correct? If so how does the received DTMF break out of the
IVR and get matched to the relevant dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo
test exten = s,5,Background(welcomemsg) exten =
s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong) exten =
6,3,Playback(tt-weasels) exten = 6,4,Wait,2 exten =
6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if
I had an entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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Re: [Asterisk-Users] Echo Cancellation (Newbie Qu)
I've tried all the other methods of dispelling the x100p echo mystery such as echo training rx tx gain through ztmon and swapping POTS lines etc etc. Can someone mail me a step by step guide to changing the echo cancellation algorithms such as Mark,Mark2, Steve etc. There isn't much to the step-by-step -- in the ztconfig.h file you will see the various echo cancellation algorithm defines: /* * Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :) */ /* #define ECHO_CAN_STEVE */ /* #define ECHO_CAN_STEVE2 */ /* #define ECHO_CAN_MARK */ #define ECHO_CAN_MARK2 /* #define ECHO_CAN_MARK3 */ In the example MARK2's defined. The general consensus is that MARK2's the best, IIRC. There is also the agressive cancellation code, which I think practically everyone wants but it breaks things like POS machines and faxes, although I think now if the driver hears a fax tone it will disable the aggressive code: /* * Uncomment for aggressive residual echo supression under * MARK2 echo canceller */ /* #define AGGRESSIVE_SUPPRESSOR */ Once you've made the changes, save 'em and make clean make make install. Regards, Andrew ___ 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] If you know your party's extension # please dial it now ...
Asterisk doesn't accept keystrokes during playback,
use BackGround to play while waiting for keystrokes.
/Stig
At 08:37 2004-03-21 -0500, you wrote:
Hi all,
I've built the usual press one for sales, 2 for support IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the options.
It seems though that one can only press one number before the IVR moves to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR)
Actually, please leave this thread on the list. Question: since this is a local connection between the Definity and Asterisk on a crossover cable, could E1 PRI be used, even though we're in the US, to realize another 8 channels? I have TN464F cards that I will be using to connect with Asterisk. Thanks, Mike James Coberly wrote: Jeb, Do you know what slot it is in? Carrier A (top) or B (bottom)? We should take this off list though and reply to me directly from this point, since this is not really * related now. There are 2 ways to do this: At the system propmt type: list configuration ds1 (will list all DS boards in the system) list configuration all will give you all boards in the system. FInd the one related to the slot you are connected to. Or if you have a restricted shell: You can look at the back of the unit, locate the amphenol you connected, there is a no. (slot #) Locate the card on the front of the unit in that slot. Should be marked TNXXX 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 -- Michael Welter Introspect Consulting, Inc. 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
RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
But if you r using AGI scripting then u can the DTMF during the
Playbacks.
e.g. $ret=$AGI-stream_file($file,12*);
here it will return 0 if nothing out of 12* pressed duringthe playback,
otherwise it will stop playing and return either 1 2 *
Regards
Azher
---
http://www.consulttech.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stig
Andersson
Sent: Sunday, March 21, 2004 7:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] If you know your party's extension #
please dial it now ...
Asterisk doesn't accept keystrokes during playback,
use BackGround to play while waiting for keystrokes.
/Stig
At 08:37 2004-03-21 -0500, you wrote:
Hi all,
I've built the usual press one for sales, 2 for support IVR which
works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the
options.
It seems though that one can only press one number before the IVR moves
to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have
any
menu choices beginning with 3 or 4. Would this be correct? If so how
does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
On Sun, 2004-03-21 at 08:56, Azher Amin wrote: But if you r using AGI scripting then u can the DTMF during the Playbacks. e.g. $ret=$AGI-stream_file($file,12*); here it will return 0 if nothing out of 12* pressed duringthe playback, otherwise it will stop playing and return either 1 2 * That's not running the Playback application. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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 T1 problem (Avaya - asterisk IVR)
Yes. Just because US carriers don't offer E1 doesn't mean we can't between our equipment in our setups. For hardware to hardware it is beneficial to utilize E1 for that exact reason. As long as your hardware on both ends supports it, no problem. James- - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 21, 2004 9:40 AM Subject: Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR) Actually, please leave this thread on the list. Question: since this is a local connection between the Definity and Asterisk on a crossover cable, could E1 PRI be used, even though we're in the US, to realize another 8 channels? I have TN464F cards that I will be using to connect with Asterisk. Thanks, Mike James Coberly wrote: Jeb, Do you know what slot it is in? Carrier A (top) or B (bottom)? We should take this off list though and reply to me directly from this point, since this is not really * related now. There are 2 ways to do this: At the system propmt type: list configuration ds1 (will list all DS boards in the system) list configuration all will give you all boards in the system. FInd the one related to the slot you are connected to. Or if you have a restricted shell: You can look at the back of the unit, locate the amphenol you connected, there is a no. (slot #) Locate the card on the front of the unit in that slot. Should be marked TNXXX 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 -- Michael Welter Introspect Consulting, Inc. 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
Re: [Asterisk-Users] Snom 200
Thanks to All who replied I have tried all the steps above. and from the website given I have two snom 200 next to each other 4403 and 4405 when I dial 4405 - 4403 nothing rings and * CLI reports voicemail/default/4403/busy when I dial 4403 - 4405 nothing rings and * CLI reports vm-theperson ... vm-isonphone If I pickup the handset I hear the dialtone I dial 13 from extensions.conf ;# Say Current Date and Time exten = 13,1,SayUnixTime(now,QABDY 'at' IMP) exten = 13,2,Wait(1) exten = 13,3,SayUnixTime(now,QABDY 'at' IMP) exyen = 13,4,Hangup * CLI report the Date and Time being said, Yet the Handset is silent, nothing coming through at all ? Where am I going wrong ? Thanks to all Barry ___ 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] UK PSTN and x100p
Hey All, I am using an x100p on a UK Telewest phone line and appear to be having problems with end user hang ups. If I call my * from and phone line and let * pick it up when I hang up the mobile or whatever I am calling from * continues with the call as if I haven't hung up. Was wondering if anyone else has had this problem and knows a way around it. Thanks, Dee ___ 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 i do voice chat without using the hardware
Hi, Yes.. i installed iaxComm in the same machine.Hope that was a wrong method. How can i uninstall iaxComm so that i can get the CLI prompt? Please help me to provide a solution for this. Thanks Regards, SurMichael Van Donselaar [EMAIL PROTECTED] wrote: On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar <[EMAIL PROTECTED]>wrote:Hi,Thanks a lot for your help.After installing iaxComm, When I test Asterisk typing# asterisk cAre you running iaxComm on the same machine as asterisk? You can't do that. I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (Nosuch file or directory) Why i am getting this error? How can i tackle thiserror?Before installing the iaxComm, i will get the CLIprompt. Now it's not getting it. So please help me tosolve this problem.Thanks p; Regards,Sur--- Michael Van Donselaar <[EMAIL PROTECTED]>wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar <[EMAIL PROTECTED]> wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur -- - Matt Ammerman <[EMAIL PROTECTED]>wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connect ions, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __Do you Yahoo!?Yahoo! Finance Tax Center - File online. File on time.http://taxes.yahoo.com/filing.html___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-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersDo you Yahoo!? Yahoo! Finance Tax Center - File online. File on time.
Re: [Asterisk-Users] T100P T1 problem (Avaya - asterisk IVR)
Michael Welter wrote: Actually, please leave this thread on the list. Question: since this is a local connection between the Definity and Asterisk on a crossover cable, could E1 PRI be used, even though we're in the US, to realize another 8 channels? I have TN464F cards that I will be using to connect with Asterisk. Thanks, Mike James Coberly wrote: Jeb, Do you know what slot it is in? Carrier A (top) or B (bottom)? We should take this off list though and reply to me directly from this point, since this is not really * related now. There are 2 ways to do this: At the system propmt type: list configuration ds1 (will list all DS boards in the system) list configuration all will give you all boards in the system. FInd the one related to the slot you are connected to. Or if you have a restricted shell: You can look at the back of the unit, locate the amphenol you connected, there is a no. (slot #) Locate the card on the front of the unit in that slot. Should be marked TNXXX James- Yes, leave on list, or someone cc me, have exact same project definity TN767E - * coming up very soon so like to follow progress. ___ 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] PRI issues with TE410P
Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes at immediate load of around 30 more calls ... I get the following processes in the ps -ax, and asterisk starts droping the calls, irq misses rise and console shows lot of pri errors (which donot occur in a smooth load of around 50 calls). Can someone explain why this happens .. however these get cleared once the channels are handled. 20992 pts/3S 0:01 zttool 21412 ?S 0:00 asterisk 21413 ?R 0:00 asterisk 21418 ?S 0:00 asterisk 21419 ?S 0:00 asterisk 21420 ?S 0:00 asterisk 21421 ?S 0:00 asterisk 21422 ?S 0:00 asterisk 21423 ?S 0:00 asterisk 21424 ?S 0:00 asterisk 21425 ?S 0:00 asterisk 21426 ?S 0:00 asterisk 21427 ?S 0:00 asterisk 21429 ?S 0:00 asterisk 21430 ?S 0:00 asterisk 21431 ?S 0:00 asterisk 21432 ?S 0:00 asterisk 21433 ?S 0:00 asterisk 21434 ?S 0:00 asterisk 21435 ?S 0:00 asterisk 21436 ?S 0:00 asterisk 21437 ?S 0:00 asterisk 21438 ?S 0:00 asterisk 21439 ?S 0:00 asterisk 21440 ?S 0:00 asterisk 21441 ?S 0:00 asterisk 21442 ?S 0:00 asterisk 21443 ?S 0:00 asterisk 21444 ?S 0:00 asterisk 21445 ?S 0:00 asterisk 21446 ?S 0:00 asterisk 21447 ?S 0:00 asterisk 21448 ?S 0:00 asterisk 21449 ?S 0:00 asterisk 21451 ?S 0:00 asterisk 21452 ?S 0:00 asterisk 21453 ?S 0:00 asterisk 21454 ?S 0:00 asterisk 21455 ?S 0:00 asterisk 21456 ?S 0:00 asterisk 21457 ?S 0:00 asterisk 21458 ?S 0:00 asterisk 21459 pts/2R 0:00 ps -ax 21460 ?S 0:00 asterisk Further I am also getting Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got S-frame while link down Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got S-frame while link down == D-Channel on span 1 up and certain IRQ misses like around 270 after an operation of 2-3 hours. Further when I load the T/E410p card I get this error ... Mar 22 00:33:48 VoiceOne kernel: TE410P: Launching card: 0 Mar 22 00:33:48 VoiceOne kernel: TE410P: Setting up global serial parameters Mar 22 00:33:48 VoiceOne kernel: TE410P: Timing from source 0 Mar 22 00:33:48 VoiceOne kernel: Found a Wildcard: Wildcard TE410P-Xilinx Mar 22 00:33:48 VoiceOne kernel: Registered tone zone 0 (United States / North America) Mar 22 00:33:48 VoiceOne kernel: TE410P: Span 1 configured for CCS/HDB3/CRC4 Mar 22 00:33:48 VoiceOne kernel: SPAN 1: Primary Sync Source Mar 22 00:33:48 VoiceOne kernel: Uhhuh. NMI received for unknown reason 20. Mar 22 00:33:48 VoiceOne kernel: Dazed and confused, but trying to continue Mar 22 00:33:48 VoiceOne kernel: Do you have a strange power saving mode enabled? Mar 22 00:33:48 VoiceOne kernel: TE410P: Span 2 configured for CCS/HDB3/CRC4 Mar 22 00:33:48 VoiceOne kernel: TE410P: Span 3 configured for CCS/HDB3/CRC4 Mar 22 00:33:48 VoiceOne kernel: TE410P: Span 4 configured for CCS/HDB3/CRC4 I have another card and above also come with that. I look forward for resolution on these issues with some suggestions to improve the stability. Coz otherwise system is not stable to be sold to client. Regards Azher Amin --- http://www.consulttech.com.pk ___ 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] Snom 200
Barry, My snom are on the same LAN as asterisk hence ... Now, you can set parameters etc. through the web interface. On the LAN where the snon is/are type in teh IP address in a browser, e.g: http://192.168.1.101 This opens the Web Interface Look in SIP Lines You will get an indication whether the phone (line) is registered. Also, for each line there is a 'Mailbox' entry, which should be the extension to check your mail at. In my case that would be '2999' which rings through to VoiceMailMain. In your case it looks like at least one of the phones thinks its extension is '4405'. That is also the default outgoin gline, i.e. asterisk sees the call as coming from '4405'. Now, unless you get a line (in the snom) setup to respond to 4401 or 4403, I don't see how they could be getting any incoming calls at all. Cheers, WW - Original Message Follows - From: [EMAIL PROTECTED] Please include the sip.conf entry for the phone you have .. SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 192.168.0.15 externip = 24.73.215.62 localnet = 192.168.0.0 localmask = 255.255.255.0 tos = lowdelay disallow = all allow = ulaw allow = all context = INVALID [4403] type= friend username= 4403 secret = 1234 nat = yes host= dynamic context = toll-access accountcode = barry mailbox = 4403 [4401] type= friend username= 4401 secret = 1234 nat = yes host= dynamic context = local-access accountcode = mark mailbox = 4401 Also, from your comments I assume that the snom 200 is on the same LAN as the [*] box? No they are not on the same LAN On the snom web interface, does it show that line 1 (which I assume you are using) is 'registered'? Not sure where you see this, First page has Outgoing line: [EMAIL PROTECTED] Sip Line Pages has Name: Phone1 Account: 4405 Registrar: 24.73.215.62 Mailbox: 4405Ringer: Ringer2 For some reason MWI, wants to dial [EMAIL PROTECTED], I have not exten or account asterisk ???, can't even find where this is set ? Thanks again Barry ___ 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 Willy Wouters ypOne Publishing ___ 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] Snom 200
Yeah, as in my reply to yoru earlier message, I don't see '4405' in your sip.conf WW - Original Message Follows - Here's another funny * CLI puts put -- Registered SIP '4405' at IP.address Port 5060 Expires 3600 and within seconds the snomm 200 beeps the MWI goes on the LCD and the light flashes a call from asterisk Not Found Willy if you could let me see you sip and config files, if you have yours working? I'm very sure it is not a LAN issue, but a config issue thanks in advance Barry ___ 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 Willy Wouters ypOne Publishing ___ 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] PRI issues with TE410P
Hello Azher- I have a similar setup in hardware, ie: TE410P running on dual-xeon system, however I'm running IVR only. I start getting the I-frame errors above about 80 simultaneous calls. I do not get IRQ misses at all. Also I do not get the startup error messages. The errors I get the most under load are the frame retransmission messages in /var/log/asterisk/messages. Do you get those as well? Since you are getting IRQ misses, you may have some basic problem, ie: something keeping the zaptel driver from getting around to servicing the TE410 interrupts. I think that they have to be serviced without fail every 1 msec, or errors start occurring. Which kernel are you running? What does your Perl script do? Did you try disabling mysql logging to lower the disk load? Maybe the disk driver is interfering with the zaptel driver interrupts. Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin Sent: Sunday, March 21, 2004 3:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI issues with TE410P Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes at immediate load of around 30 more calls ... I get the following processes in the ps -ax, and asterisk starts droping the calls, irq misses rise and console shows lot of pri errors (which donot occur in a smooth load of around 50 calls). Can someone explain why this happens .. however these get cleared once the channels are handled. 20992 pts/3S 0:01 zttool 21412 ?S 0:00 asterisk 21413 ?R 0:00 asterisk 21418 ?S 0:00 asterisk 21419 ?S 0:00 asterisk 21420 ?S 0:00 asterisk 21421 ?S 0:00 asterisk 21422 ?S 0:00 asterisk 21423 ?S 0:00 asterisk 21424 ?S 0:00 asterisk 21425 ?S 0:00 asterisk 21426 ?S 0:00 asterisk 21427 ?S 0:00 asterisk 21429 ?S 0:00 asterisk 21430 ?S 0:00 asterisk 21431 ?S 0:00 asterisk 21432 ?S 0:00 asterisk 21433 ?S 0:00 asterisk 21434 ?S 0:00 asterisk 21435 ?S 0:00 asterisk 21436 ?S 0:00 asterisk 21437 ?S 0:00 asterisk 21438 ?S 0:00 asterisk 21439 ?S 0:00 asterisk 21440 ?S 0:00 asterisk 21441 ?S 0:00 asterisk 21442 ?S 0:00 asterisk 21443 ?S 0:00 asterisk 21444 ?S 0:00 asterisk 21445 ?S 0:00 asterisk 21446 ?S 0:00 asterisk 21447 ?S 0:00 asterisk 21448 ?S 0:00 asterisk 21449 ?S 0:00 asterisk 21451 ?S 0:00 asterisk 21452 ?S 0:00 asterisk 21453 ?S 0:00 asterisk 21454 ?S 0:00 asterisk 21455 ?S 0:00 asterisk 21456 ?S 0:00 asterisk 21457 ?S 0:00 asterisk 21458 ?S 0:00 asterisk 21459 pts/2R 0:00 ps -ax 21460 ?S 0:00 asterisk Further I am also getting Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got I-frame while link state 2 Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got S-frame while link down Mar 22 01:33:09 WARNING[114696]: chan_zap.c:5970 zt_pri_error: PRI: !! Got S-frame while link down == D-Channel on span 1 up and certain IRQ misses like around 270 after an operation of 2-3 hours. Further when I load the T/E410p card I get this error ... Mar 22 00:33:48 VoiceOne kernel: TE410P: Launching card: 0 Mar 22 00:33:48 VoiceOne kernel: TE410P: Setting up global serial parameters Mar 22 00:33:48 VoiceOne kernel: TE410P: Timing from source 0 Mar 22 00:33:48 VoiceOne kernel: Found a Wildcard: Wildcard TE410P-Xilinx Mar 22 00:33:48 VoiceOne kernel: Registered tone zone 0 (United States / North America) Mar 22 00:33:48 VoiceOne kernel: TE410P: Span 1 configured for CCS/HDB3/CRC4 Mar 22 00:33:48 VoiceOne kernel: SPAN 1: Primary Sync Source Mar 22 00:33:48
Re: [Asterisk-Users] Snom 200
Thanks Willy and others
It seems I am not able to make myself clear on my two problems
I wish to try again, as I'm sure it is not the phones, but
some stupid config problem on my part. I repeat alot of
what I have said in order to try make myself clear.
I have a 4401, 4403 and 4405 in sip.conf all set the same
[440x] {where x is either 1, 3 or 5}
type= friend
username= 440x
secret = 1234
nat = yes
host= dynamic
context = local-access
accountcode = mark
mailbox = 440x
dtmfmode= inband
I have extensions 4401, 4403 and 4405 in extensions.conf
{where x is either 1, 3 or 5}
exten = 440x,1,Dial(SIP/440x,20)
exten = 440x,2,Voicemail2(u${EXTEN})
exten = 440x,3,Hangup
exten = 440x,102,Voicemail2(b${EXTEN})
exten = 440x,103,Hangup
the Asrerisk CLI, reports everything ok, in that the two phones are
registered see here
* CLI
-- Registered SIP '4405' at 24.129.a.b port 15061 expires 3600 (I
assume the port diff, due to two
phones
on the same network, my guess
I never
set ports anywhere)
-- Registered SIP '4403' at 24.129.a.b port 5060 expires 3600
* CLI
Two of the phones 4403 and 4405 are configured the same via the web browser.
So I have the phones configured the same, I have * deal with the phones the
same (as the * configs are the same). Both Phones have Version 2.03o 5442.
So seeing that phone on both sides (actual phone and * server) are
configured the same, you would expect them to act the same. Not the case.
If 4403 dials 4405 * CLI reports this
== Everyone is busy at this time
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'vm-isonphone' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
If 4405 dials 4403 * CLI reports this
== Everyone is busy at this time
-- Playing 'voicemail/default/4403/busy' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
The two phones are next to each other.
When 4403 dials 4405, 4405 does not even ring or anything
When 4405 dials 4403, 4403 does not even ring or anything
Yet as from above The Phones are reported busy, busy by who
they are actually idle.
That is all problem 1, the phones report busy, while sitting idle
Problem 2, If I pick up the handset I hear the dialtone (proof the phone is
connected)
When I dial an extension which is set to play the time and date, the * CLI
scrolls the
voice saying date an time. Yet the Handset is silent, Why? If I hear the
dialtone at the
start why does the handset go dead, surely I should hear the voice on the
other side
talking (in this case the * server)?
See this post is long and so I have not posted all sip.conf and
extensions.conf file, just parts
If needed I can e-mail direct the sip.conf and extensions.conf.
Thanks again,
Barry (Just trying to get my new snom 200s to work)
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[Asterisk-Users] Mantis - closing feature request when feature no added
Ok, so I've re-reported a feature request http://bugs.digium.com/bug_view_page.php?bug_id=0001265 because http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9 was closed for no apparent reason. Is it now policy to simply close off feature requests when they haven't been added? If it is now policy please let us know so that we can save everyone a lot of time by not bothering to add feature requests in the first place... Andy ___ 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] Snom 200 Voice Call / Paging
To All, Several months (2003) ago there was a discussion regarding overhead paging intercom functionality with SIP / Asterisk. Jerry Gibson, John Todd and various others participated (from checking the archives). One person even responded that they had the stuff working with the snom 200s. Voice Call (i.e. on-hook speaker/mic) is realy important in a lot of apps. It would appear that the snom 200 and by extension the snom 105 support the functionality. I will be happy to make a wiki entry to explain demo this functionality once I have it working properly. I also understand that the (mis)use of conferencing is frowned upon as it wastes bandwidth and CPU. However, until a better way comes around, that is not a problem as there are quite a few applications where (a) one needs Voice Call (which is 1 - 1) and / or an 'allPage' which can be limited to a subset of all phones. Typically phones which are in designated or public areas, conference rooms, etc. The BW/CPU issue can be controlled. Better a limited solution than no solution at all ;) I am also allowing for the limitation that all participating phones are on the same LAN with the [*]. Anyone who has this successfully working with snom, please respond .. Using the [*] sound card for a separate PA system is NOT an option ;) As I said, I will be 'distilling' the info and turn it into a wiki entry. Cheers and TIA, Willy Willy Wouters ypOne Publishing ___ 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 i do voice chat without using the hardware
Hi, Yes... finally i solved that problem. I am getting CLI prompt. When i type asterisk -r command, Now i got display as [EMAIL PROTECTED] asterisk]# asterisk -r Asterisk CVS-03/18/04-18:01:45, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-03/18/04-18:01:45 currently running on edventure17 (pid = 3650) edventure17*CLI I would like to get some help from you. My server ip is 192.168.1.1 and i would like to connect to another ip 192.168.1.2. So how can i specify the ip 192.168.1.2 so that make a call from 192.168.1.1? Should i install softphone s/w in server (192.168.1.1) and other machine (192.168.1.2)? In sip.conf file how can i specify the ip 102.168.1.2 If you have time, please help me to get a solution. Thanks Regards, Suresh --- Girish Gopinath [EMAIL PROTECTED] wrote: Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can i do voice chat without using the hardware Date: Fri, 19 Mar 2004 05:50:00 -0800 (PST) Thanks a lot for your valuable information. I will go through it once again. Still i don't have any idea to connect two PC's. Hope i may get help from you. For configuring 2 softphones with Asterisk see this link: http://www.automated.it/guidetoasterisk.htm That helped me a lot in learning Asterisk. It explains configuring your sip phones with Asterisk. Is there any softwares like X-lite for Linux? Yes, I think you can use linophone. But i was not able to install linophone because of some make issues. Also i have tested the softphone from zultys. It works well with Asterisk. You can get it from their web site:http://www.zultys.com Regards, Girish _ Catch the formula fever! Get all the latest news. http://www.msn.co.in/formula2004/ Right here on MSN. ___ 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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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] UK PSTN and x100p
Hi Iain I use telewest my self and i have it set up to use Kewlstart, it does disconnect the call, but its only after the teleewest line plays a ringing noise, and then the telewest woman says the other person has cleared. HTH Welby Iain Stevenson wrote: Well. if the Telewest line signalling is the same as BT uses it should work. When the call ends the Telewest switch should signal this with a change in the line power which the X100P relies on to disconnect. the call. You'll probably need to measure the line voltage to sort this out. If you have access to a BT line it's worth trying the X100P on that. Iain --On Sunday, March 21, 2004 15:32:54 + Dee Lowndes [EMAIL PROTECTED] wrote: Hey All, I am using an x100p on a UK Telewest phone line and appear to be having problems with end user hang ups. If I call my * from and phone line and let * pick it up when I hang up the mobile or whatever I am calling from * continues with the call as if I haven't hung up. Was wondering if anyone else has had this problem and knows a way around it. Thanks, Dee ___ 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] Sound prompt conversion utility?
Does anybody know of a utility that can convert voice prompts from one codec to another? I'm trying to convert some prompts stored as .gsm to .g729 - Khan ___ 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] Any Polycom Experts Out There?
Hi, all We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work very well and we're quite happy with them. They register fine and all four are able to place and receive calls, BUT two of them are behind NAT routers and when they place a call on hold, the call is dropped within 5 seconds. I couldn't find any relevant items in the archive search using terms like SIP, NAT and HOLD. The server has a public static IP, two phones have public static IPs and the two with the hold problem have dynamic NAT'ed IPs. Not sure what other info might be helpful. Any pointers in the right direction would be appreciated. -russ ___ 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] Any Polycom Experts Out There?
We're using Asterisk CVS 3-19-04 with four polycom IP600s. The work very well and we're quite happy with them. They register fine and all four are able to place and receive calls, BUT two of them are behind NAT routers and when they place a call on hold, the call is dropped within 5 seconds. I couldn't find any relevant items in the archive search using terms like SIP, NAT and HOLD. The server has a public static IP, two phones have public static IPs and the two with the hold problem have dynamic NAT'ed IPs. Not sure what other info might be helpful. I'm no Polycom expert, but there's a nat section in your Polycom XML config files that you might tinker with. There's a brief section about it in the Polycom manual. And that's the limit of my knowledge Carey ___ 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 i do voice chat without using the hardware
On Sun, 21 Mar 2004 07:38:17 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: Hi, Yes.. i installed iaxComm in the same machine. Hope that was a wrong method. How can i uninstall iaxComm so that i can get the CLI prompt? Please help me to provide a solution for this. Thanks Regards, Sur You don't need to. From looking at another of your posts, it looks like you've got asterisk running in the background. Typing asterisk -r should get you the CLI of the asterisk that is running in the backgound. Michael Van Donselaar [EMAIL PROTECTED] wrote: On Sun, 21 Mar 2004 04:00:39 -0800 (PST), suresh kumar wrote: Hi, Thanks a lot for your help. After installing iaxComm, When I test Asterisk typing # asterisk c Are you running iaxComm on the same machine as asterisk? You can't do that. I got a display like this (Not getting any CLI prompt) [chan_iax.so] = (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory ) Why i am getting this error? How can i tackle this error? Before installing the iaxComm, i will get the CLI prompt. Now it's not getting it. So please help me to solve this problem. Thanks Regards, Sur --- Michael Van Donselaar wrote: On Fri, 19 Mar 2004 05:53:44 -0800 (PST), suresh kumar wrote: Hi, Thanks for your help. I had gone through the www.voip-info.org and got more information regarding the asterisk. Still now i am not clear, how can i test this software. I had gone through the mailarchieves, but didn't get any solution. My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? iaxComm uses asterisk's native IAX protocol. It runs on Windows, Linux and OSX. Precompiled binaries for RedHat 9, Windows, and OSX (Panther) ara available at: http://iaxclient.sourceforge.net/iaxcomm/index.html linphone is a SIP softphone for Linux: http://www.linphone.org I am expecting an help from experienced person like you. Or can you please send me the link where i can get more information to tackle my problem. Thanking you, Best Regards, Sur --- Matt Ammerman wrote: Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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 Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. ___ 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] If you know your party's extension # please dial it now ...
If Asterisk can't determine whether you want 3 or 3XXX, it will wait for
DigitTimeout. So if someone dials 3 for echo test, it will take 3
seconds in your case before it jumps to that extension.
David
Mark Phillips wrote:
Hi all,
I've built the usual press one for sales, 2 for support IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the options.
It seems though that one can only press one number before the IVR moves to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
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Re: [Asterisk-Users] Snom 200
Barry Fawthrop wrote: Thanks Willy and others It seems I am not able to make myself clear on my two problems I wish to try again, as I'm sure it is not the phones, but some stupid config problem on my part. I repeat alot of what I have said in order to try make myself clear. SNIP That is all problem 1, the phones report busy, while sitting idle This is a known problem. Sometimes the SNOM's seem to go to BUSY without any cause. At least no indicator is shown in the display. A possible solution is posted here: http://www.voip-info.org/wiki-SNOM+phones If it does not solve the problem, try to reset the phone to default settings and power cycle it. With the latest firmware I did not see the problem yet, but I see the other bugs (crash + transfer, I'm busy emailing with SNOM about these bugs). Problem 2, If I pick up the handset I hear the dialtone (proof the phone is connected) When I dial an extension which is set to play the time and date, the * CLI scrolls the voice saying date an time. Yet the Handset is silent, Why? If I hear the dialtone at the start why does the handset go dead, surely I should hear the voice on the other side talking (in this case the * server)? The dialtone is no indication you can setup an RTP stream. You should test it with the asterisk built in ECHO server. Make sure there is no firewall activated which can block the traffic. If it still doesn't work, check the sip debug output for errors or retransmits. Kind regards, Geert ___ 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 i do voice chat without using the hardware
On Sun, 21 Mar 2004 09:11:33 -0800 (PST), suresh kumar [EMAIL PROTECTED] wrote: snip I would like to get some help from you. My server ip is 192.168.1.1 and i would like to connect to another ip 192.168.1.2. So how can i specify the ip 192.168.1.2 so that make a call from 192.168.1.1? Basic configuration is described in the QUICKSTART that came with the binary. But, since you want to originate calls from the asterisk server, that's a bit different. I'm assuming that you have asterisk installed and working with your sound card. I'm also assuming that you still have the default extensions.conf. If so, you should be able to type dial 600 at the CLI prompt and get the echo test. If not, you'll have to get that fixed before going further. If you can dial extensions from the console OK, then just 1. Make and iax.conf entry for an extension [101] type=friend host=dynamic secret=foo context=default callerid=Remote PC 101 diasallow=all allow=gsm 2. Make an extensions.conf entry for that extension in the default context exten = 101,1,Dial(IAX2/101) 3. Configure iaxComm on the other machine to use the iaxconf entry (username 101, password foo) as described in the QUICKSTART. Should i install softphone s/w in server (192.168.1.1) and other machine (192.168.1.2)? You don't want iaxComm installed on the asterisk server. Just on the remote machines. In sip.conf file how can i specify the ip 102.168.1.2 iaxComm does not use the SIP protocol. It's config file is iax.conf If you have time, please help me to get a solution. Thanks Regards, Suresh --- Girish Gopinath [EMAIL PROTECTED] wrote: Suresh, From: suresh kumar [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can i do voice chat without using the hardware Date: Fri, 19 Mar 2004 05:50:00 -0800 (PST) Thanks a lot for your valuable information. I will go through it once again. Still i don't have any idea to connect two PC's. Hope i may get help from you. For configuring 2 softphones with Asterisk see this link: http://www.automated.it/guidetoasterisk.htm That helped me a lot in learning Asterisk. It explains configuring your sip phones with Asterisk. Is there any softwares like X-lite for Linux? Yes, I think you can use linophone. But i was not able to install linophone because of some make issues. Also i have tested the softphone from zultys. It works well with Asterisk. You can get it from their web site:http://www.zultys.com Regards, Girish _ Catch the formula fever! Get all the latest news. http://www.msn.co.in/formula2004/ Right here on MSN. ___ 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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ 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] Discriminate on IAXTEL dial-in
On Sun, Mar 21, 2004 at 07:42:05AM -0600, Eric Wieling wrote: You are correct, IAXtel does not send the called number. Calls from both IAXTel accounts will fall into the s extension. Oh, I see. So I have now implemented a menu. If you call 1-700-895-5211 you can now dial 0800 numbers in Switzerland (dial 00800 800 800 for Swisscom fixnet for example). If you dial +41 328 41 47 74 you get the other way around, ie dial into IAX. And I am extension 200 (or 9). I however have another question: - apparently when I call from ISDN to an IAX gnophone, I get a very short ring then an error: (XXX are mine) -- Calling using options 'exten=s;callerid=03284140XX;language=en;formats=2;capability=65283;version=1;adsicpe=0' -- Called XXX -- Call accepted by 80.83.50.XXX (format GSM) -- Format for call is GSM -- IAX[XXX]/50 is ringing Mar 21 20:43:29 DEBUG[33810]: channel.c:1265 ast_indicate: Driver for channel 'CAPI[contr4/8414774]/10' does not support indication 3, emulating it Mar 21 20:43:29 ERROR[33810]: chan_capi.c:851 capi_write: not a voice frame Mar 21 20:43:29 WARNING[33810]: app_dial.c:313 wait_for_answer: Unable to forward image Mar 21 20:43:29 DEBUG[33810]: chan_iax.c:1861 iax_hangup: We're hanging up IAX[XXX]/50 now... -- Hungup 'IAX[XXX]/50' == Spawn extension (macro-dial-extension, s, 3) exited non-zero on 'CAPI[contr4/8414774]/10' in macro 'dial-extension' - the problem doesn't happen when calling from a SIP phone. - the problem also happens if you do ISDN - IAX - IAX gnoèphone. Probably this is a bug in chan_capi-0.3.0 ? ___ 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't get the full callerid php/agi
Hi David, Thanks, yes that was the problem. Really appreciate your tip. Cheers Sathya From: David Croft [EMAIL PROTECTED] Organization: Sargasso Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] can't get the full callerid php/agi Reply-To: [EMAIL PROTECTED] Your script is receiving the data correctly, as you will see if you actually dump that data to a file rather than back to the asterisk console. The problem is actually in your VERBOSE statement. You are passing back this string: VERBOSE Sathya Weerasooriya 1001 Naturally asterisk is confused by this quote nesting. Try this line instead: echo VERBOSE \.str_replace(\, \\\, $temp).\\n; David ___ 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] Use of Alert_Info with C7960?
The wiki indicates Alert_Info can be set to a number, and implies that number is the ringer type listed on the phone. Is there a way to select one of the internal ringer types via Alert_Info? My understanding is that: 1. 7940/7960 pre version 6 may support numeric values 1-5 (not tested). 2. 7940/7960 firmware version 6 supports textual ALERT_INFO (bellcore-dr2) etc. (see version 6.0 release notes) 3. You cannot specify the ringtone to use, only what I guess I'd call the 'cadence' - you'll notice dr1 through dr5 ring in different patterns. 4. Your current ringtone is used with the specified cadence. The cadences are mostly so similar as to be useless so I have resorted to having the 7960s register multiple line appearances so you can see which one is ringing through, rather than using distinctive ring. Yes, I did the same thing months ago. If anyone has successfully got a custom ring tone, do chime in. Tried lots of different approaches and the only ones that actually work are the bellcore examples that others have stated. Similarly, if you know how to get VXML_URL to work on the 7960, let me know. This just appends stuff to the To: SIP header. I see no mention of this (or XML push) anywhere in the Cisco documentation, so I'm disinclined to believe the wiki/source that this field is actually for the Cisco phones. Maybe something else. I too have played around with VXML and could not find any support for this whatsoever. I'd guess it might be related to Cisco's non-sip software. Rich ___ 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 installation problem
OK, I solved the problem by myself: openssl-devel was not installed. Unfortunately, there's not a deb-package, so I had to convert the RPM. Regards, thomas On 21 Mar 2004 at 21:29, Thomas Schroeter wrote: Hello, I have the following problem installing Asterias on Debian woody: Installation of zaptel and libpri works find, after make clean; make install; for asterisk, it exits with make: *** [ast_expr.c] Error 1 Before there were several errors, starting with: cli.c:31: build.h: No such file or directory dlfcn.c:40: mach-o/dyld.h: No such file or directory dlfcn.c:41: mach-o/nlist.h: No such file or directory dlfcn.c:42: mach-o/getsect.h: No such file or directory What's the probem...? Regards, Thomas --- Thomas Schroeter // +49-175-4624147 // +49-40-72976451 ___ 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 --- Thomas Schroeter // +49-175-4624147 // +49-40-72976451 ___ 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] Discriminate on IAXTEL dial-in
- apparently when I call from ISDN to an IAX gnophone, I get a very short ring then an error: (XXX are mine) This doesn't happen when gnophone is configured as `Use Asterisk' apparently. So this is now solved. ___ 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] Echo Cancellation (Newbie Qu)
I've tried all the other methods of dispelling the x100p echo mystery such as echo training rx tx gain through ztmon and swapping POTS lines etc etc. Can someone mail me a step by step guide to changing the echo cancellation algorithms such as Mark,Mark2, Steve etc. I spent a fair amount of time back in the October/November timeframe mucking around with echo problems. I had substantial issues with it at that time. Mark made some changes based on lots of complaints back in that timeframe. One of the things that is not at all clear (in postings and other doc) is that changes made to the zapata.conf file must be followed by a total restart of asterisk. That includes changes to rxgain, txgain, etc. A reload does not cause the x100p driver to re-read zapata.conf. I've been running with Dec 4th version of the zapata CVS and the following x100p config, and echo is just barely perceptable during the first half second or so of a call. context=inbound-home switchtype=national signalling=fxs_ks echotraining=yes usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=no threewaycalling=no echocancel=yes echocancelwhenbridged=yes rxgain=-0.0 txgain=-0.0 callgroup=2 immediate=no callprogress=no musiconhold=default channel = 1 Rich ___ 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] If you know your party's extension # please dial it now ...
I've built the usual press one for sales, 2 for support IVR which works
fine but I'm having difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the options.
It seems though that one can only press one number before the IVR moves to
the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have any
menu choices beginning with 3 or 4. Would this be correct? If so how does
the received DTMF break out of the IVR and get matched to the relevant
dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
Mark,
Here's a partial copy of my ivr, and I too am using the 3xxx extensions.
Notice I avoided use of option 3 in the ivr menues.
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,20
exten = s,5,Background(npi-greeting) ; Thanks for calling press 1 for
exten = 1,1,Goto(local-extns|3014|1) ; Sales
exten = 2,1,Dial(${PHONE1}${PHONE2},15) ; Technical Services
exten = 2,2,Voicemail2(u3000)
exten = 2,102,Voicemail2(b3000)
exten = 2,103,Hangup
exten = 8,1,Goto(npilist|s|1); Company directory list
exten = 9,1,Goto(npitest|s|1); VoIP Testing Menu
Rich
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RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
You don't have to avoid using an option 3 when even if extensions are
3XXX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Sunday, March 21, 2004 4:19 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] If you know your party's
extension # please dial it now ...
I've built the usual press one for sales, 2 for support IVR which
works fine but I'm having difficulty in allowing callers to type in
whole extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted
the IVR below
(just in case someone wants one). The welcome message
states callers
should type in the extension number they want or choose
from the options.
It seems though that one can only press one number before the IVR
moves to the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I
can't have
any menu choices beginning with 3 or 4. Would this be
correct? If so
how does the received DTMF break out of the IVR and get
matched to the
relevant dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten = s,5,Background(welcomemsg) exten =
s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
Mark,
Here's a partial copy of my ivr, and I too am using the 3xxx
extensions.
Notice I avoided use of option 3 in the ivr menues.
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,20
exten = s,5,Background(npi-greeting) ; Thanks for calling
press 1 for
exten = 1,1,Goto(local-extns|3014|1) ; Sales exten =
2,1,Dial(${PHONE1}${PHONE2},15) ; Technical Services exten
= 2,2,Voicemail2(u3000) exten = 2,102,Voicemail2(b3000)
exten = 2,103,Hangup
exten = 8,1,Goto(npilist|s|1); Company directory list
exten = 9,1,Goto(npitest|s|1); VoIP Testing Menu
Rich
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[Asterisk-Users] AGI startup on channel when asterisk starts
All, I am looking for a way to have my AGI startup on a channel automatically when asterisk starts. Is this possible? I have my AGI working for when a call comes in - however I would like the AGI started up automatically with asterisk on a couple channels as I want to monitor my database and when things happen place a couple calls etc. I am aware of the outgoing directory but that is not exactly what I am wanting to do unless there is a way to have the commands in the file not actually dial but just give me a channel and start my AGI. They my AGI can place the call if that is what is required. Jerry ___ 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] If you know your party's extension # please dial it now ...
Please elaborate ...
- Original Message Follows -
You don't have to avoid using an option 3 when even if
extensions are 3XXX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Rich Adamson
Sent: Sunday, March 21, 2004 4:19 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] If you know your party's
extension # please dial it now ...
I've built the usual press one for sales, 2 for
support IVR which works fine but I'm having
difficulty in allowing callers to type in whole
extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have
pasted the IVR below
(just in case someone wants one). The welcome message
states callers
should type in the extension number they want or
choose from the options.
It seems though that one can only press one number
before the IVR moves to the next step.
I'm starting to think that if my extn's are 3xxx and
4xxx I can't have
any menu choices beginning with 3 or 4. Would this be
correct? If so
how does the received DTMF break out of the IVR and
get matched to the
relevant dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3
for echo test exten = s,5,Background(welcomemsg)
exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
Mark,
Here's a partial copy of my ivr, and I too am using the
3xxx extensions.
Notice I avoided use of option 3 in the ivr menues.
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,20
exten = s,5,Background(npi-greeting) ; Thanks for
calling press 1 for
exten = 1,1,Goto(local-extns|3014|1) ; Sales exten =
2,1,Dial(${PHONE1}${PHONE2},15) ; Technical Services
exten = 2,2,Voicemail2(u3000) exten = 2,102
,Voicemail2(b3000) exten = 2,103,Hangup
exten = 8,1,Goto(npilist|s|1); Company
directory list exten = 9,1,Goto(npitest|s|1);
VoIP Testing Menu
Rich
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Re: [Asterisk-Users] chan_sccp
Oh, it also seems to crash my Asterisk. (0.7.2). On Sun, 2004-03-21 at 16:27, Eric Wieling wrote: My Cisco 7910 works fine with chan_skinny. I'm now trying to use the 7910 with chan_sccp. The phone hangs with a message Requesting Server List. Has anyone seen this problem. Happens with both chan_sccp CVS and with 0.02. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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] UK PSTN and x100p
Hi Dee Lowndes wrote: Hi, I use telewest my self and i have it set up to use Kewlstart, it does disconnect the call, but its only after the teleewest line plays a ringing noise, and then the telewest woman says the other person has cleared. That is exactly what happens with mine by any chance did you get caller id working with it? It was a matter of pluging it in and going, It all worked here. My excahnge definatly is using the bellcore standard for caller id. If i remeber right a few of the telewest areas use BT's standard. (i'm in edinburgh, which is a DMS exchange, but other areas are using system x's, nokias, ericosons etc ( http://www.telewest.co.uk/business/customerservices/cs_userguides.html )) HTH Welby Iain Stevenson wrote: Well. if the Telewest line signalling is the same as BT uses it should work. When the call ends the Telewest switch should signal this with a change in the line power which the X100P relies on to disconnect. the call. You'll probably need to measure the line voltage to sort this out. If I find the voltage drop out can I configure the x100p to do it based on the new voltage drop. If so where and how? If you have access to a BT line it's worth trying the X100P on that. Iain No BT line unfortunately. Cheers, Dee --On Sunday, March 21, 2004 15:32:54 + Dee Lowndes [EMAIL PROTECTED] wrote: Hey All, I am using an x100p on a UK Telewest phone line and appear to be having problems with end user hang ups. If I call my * from and phone line and let * pick it up when I hang up the mobile or whatever I am calling from * continues with the call as if I haven't hung up. Was wondering if anyone else has had this problem and knows a way around it. Thanks, Dee ___ 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 ___ 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] UK - 1471
In the UK we have a service that if you dial 1471, the last 6 calls are read out to you and you can pick which one you want by pressing 3, this means that 1471 shows in the cdr, has anyone created a script or an application that will read out the last callers and then dial the number? ( that they would like to share? I only ask before I start to re invent the wheel Thanks Robb ___ 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] UK - 1471
Hi Robert Boardman wrote: In the UK we have a service that if you dial 1471, the last 6 calls are read out to you and you can pick which one you want by pressing 3, this means that 1471 shows in the cdr, has anyone created a script or an application that will read out the last callers and then dial the number? ( that they would like to share? I only ask before I start to re invent the wheel Thanks Robb I wrote a scrpit, not the best in the world but it works (the call back didnt last time i checked but it might with a bit of work). Its availbe at http://www.wheely-bin.co.uk/asterisk/ HTH Welby ___ 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] If you know your party's extension # please dial it now ...
Title: RE: [Asterisk-Users] If you know your party's extension # please dial it now ...
Simply define your local
extensions as well as your virtual extensions...
exten 1,1,Play...
exten 2,1,Play...
exten 3,1,Play...
exten 333,1,Play...
When they press 1 the system will
immediately Play, when they press 2 the system will immidiately
play.
When they press 3 the system will wait x
amount of seconds for more input because there is the 333 extension, if no more
numbers are pressed it will go to 3,1 if 333 is pressed it will play the
approrpiate file.
From: [EMAIL PROTECTED] on
behalf of [EMAIL PROTECTED]Sent: Sun 3/21/2004 5:09 PMTo:
Asterisk UsersSubject: RE: [Asterisk-Users] If you know your party's
extension # please dial it now ...
Please elaborate ...- Original Message Follows
- You don't have to avoid using an option 3 when even if
extensions are 3XXX -Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]
On Behalf Of Rich Adamson Sent: Sunday, March
21, 2004 4:19 PM To: Asterisk Users Subject: Re:
[Asterisk-Users] If you know your party's extension # please dial
it now ...I've built the usual "press one for
sales, 2 for support" IVR which works fine but I'm
having difficulty in allowing callers to type in
whole extension numbers. My internal extn
ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message
states callers should type in the extension number they want
or choose from the options. It seems though
that one can only press one number before the IVR moves
to the next step. I'm starting to think that
if my extn's are 3xxx and 4xxx I can't have
any menu choices beginning with 3 or 4. Would this be correct? If
so how does the received DTMF break out of the IVR and
get matched to the relevant dialplan entry?
[mainmenu]
exten = s,1,Answer exten =
s,2,SetMusicOnHold(default) exten =
s,3,DigitTimeout,3 exten =
s,4,ResponseTimeout,5 ;SAI menu - 1 for tech support, 2
for voicemail, 3 for echo test exten =
s,5,Background(welcomemsg) exten =
s,6,Background(choosemsg) ;
Sales exten = 1,1,Dial,SIP/3400|20 exten
= 1,2,Voicemail(3400) exten =
1,3,Goto(mainmenu,s,60 Mark,
Here's a partial copy of my ivr, and I too am using
the 3xxx extensions. Notice I avoided use of
option 3 in the ivr menues. [bus-ivr-main]
exten = s,1,Wait,1 exten = s,2,Answer
exten = s,3,DigitTimeout,5 exten =
s,4,ResponseTimeout,20 exten =
s,5,Background(npi-greeting) ; "Thanks for calling
press 1
for"
exten = 1,1,Goto(local-extns|3014|1) ; Sales exten
= 2,1,Dial(${PHONE1}${PHONE2},15) ; Technical
Services exten = 2,2,Voicemail2(u3000) exten =
2,102 ,Voicemail2(b3000) exten = 2,103,Hangup
exten = 8,1,Goto(npilist|s|1)
; Company directory list exten =
9,1,Goto(npitest|s|1) ;
VoIP Testing Menu Rich
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Re: [Asterisk-Users] Cisco 7960 vs 7905
I'm interested in picking up a Cisco SIP phone, but I don't have enough information to decide between the 7940/60 family and the 7905/12 family. Between the wiki and Cisco's web site, it seems clean that the 7905/12 don't have a speakerphone, and that the 7905 doesn't have a built-in Ethernet switch. The wiki suggests that the 7905/12 has a better SIP implementation and a higher-resolution screen, but that's about all that I can find comparing the two. Can anyone with both of them give me a bit more information? A few things that I'm interested in: - XML directory support: how many entries supported, how many lines displayed on the screen on each? - SIP Alert-Info ringtones. The 7960 can choose from the standard bellcore set right now, but not custom tones. How does the 7905 compare? - XML services. Is there a difference, or indeed any documentation anywhere? - SIP implementation quality. The wiki suggests that the 7905 works better, but with no examples. Are there actually problems with the 7960? - Lifespan. The 7960 is currently running v6.3, while the 7905 is running v1.01. Cisco seems to be be putting more work into the higher-end family. - Subjective usability. Does either one work or feel better? I only have the 7960, so can't comment much on the 7905. The 7960 is a very stable business-class phone that has high acceptability by non- techie users. Feels looks like a telephone and doesn't slide across the desk when you stretch the handset cord. Spearkerphone and all work very well. The 7940/7960 have been around for a long time while the 7905 is a rather recent addition to their product line. I believe the 7905 only supports the Cisco proprietary firmware (not sip) while the 7960 supports either Cisco or sip. That's probably why you're seeing v1 verses v6.3 or whatever. The screen on the 7960 is a rather low resolution one, and therefore does not display much data. Pressing the directory button (and selecting external directory) does use xml to look up entries from a remote web server (apache in my case), and appears to load all entries at the server at one time (therefore, there probably is some magic limit as to number of entries). Cisco did produce an xml document for the phone. The directory function is not all that useful as you need to manually scroll through the entire list to fine the entry you want. The screen displays three entries (on six lines); first line is the name while the second line is the telephone number. Apparently some of the functions that exist in the Cisco proprietary firmware do not have equivalent functions using the sip firmware (like the ring tones, services button, etc). If you buy one, I'd suggest purchasing the Cisco maintenance (about $8 per year in US) as that gives you access to a fair amount of Cisco documentation as well as software upgrades. From a personal perspective (with 20+ years in technical telephony engineering), I'd take the 7960 over the snom products any day of the week. But I can't compare it to lots of other probably fine products out there since I've not tested/played with them. ___ 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 installation problem
On Sun, Mar 21, 2004 at 10:09:33PM +0100, Thomas Schroeter wrote: OK, I solved the problem by myself: openssl-devel was not installed. Unfortunately, there's not a deb-package, so I had to convert the RPM. Here's the one I use: http://packages.debian.org/stable/devel/libssl-dev -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ 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] Cisco 7960 vs 7905
The 7905G (but not the non-G) supports SIP. It does NOT support XML. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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] Cisco 7960 vs 7905
On Sun, Mar 21, 2004 at 05:53:00PM -0600, Rich Adamson wrote: The 7940/7960 have been around for a long time while the 7905 is a rather recent addition to their product line. I believe the 7905 only supports the Cisco proprietary firmware (not sip) while the 7960 supports either Cisco or sip. That's probably why you're seeing v1 verses v6.3 or whatever. The 7905 and the 7905G both run the 1.01 SIP firmware. -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ 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] Important: The Asterisk Mailing list (newsubject)
On Fri, 2004-03-19 at 22:23, Darren Nickerson wrote: Folks, I strongly support removing the current reply-to-list setting, and you should too. Like many new list admins, I once thought the reply-to was kewel. Requests to remove it kept coming up, ... usually around the same time someone embarrassed themselves by posting a personal reply/flame to the list. Someone, in frustration, finally pointed me to the following URL: http://www.unicom.com/pw/reply-to-harmful.html I saw the light. Please can the list admin step in and end this thread by either: a) announcing that the reply-to override has been removed b) announcing their resignation ;-) I'm sorry you saw the wrong light. You are peering into a light that will anger many more of us to the point of removing ourselves from the list as it becomes impossible to filter appropriately. Reply to group has a nasty habit of piling up addresses and then people who have dropped out of the thread are still getting barraged by messages where their address is still a part of it. It is bad enough we have users too lame to click on a link to the submission url and instead just reply and erase old content, your suggestion would just make people more likely to get nailed with unrelated content. Open source software thrives by efficient and open communications. To start suggesting people take useful commentary off list by making it less easy to reply to the list only reduces our resources. It also starts a lot of private communications and possibly private flame wars. If you post embarrassing information, or if your post embarrass you in public, maybe they didn't need to be said in the first place. Your aversion to fixing a to line when you take a message off list is not worth breaking good mail filtering. You can probably blame me for the original switch of the Reply-To header. I believe I am the one who requested it soo long ago. -- 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] Cisco 7960 v6.3 firmware
FYI... Cisco released v6.3 sip firmware around March 12th. Resolved caveats: # definitions within dialplan file are not functional Wrong SDP message for a G711 codec negotiation 79x0 Config is not saved on upgrade to LA/BA as LA is named P003 ___ 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] DID with X100P?
On Fri, 2004-03-19 at 11:34, Victor Perez wrote: Is there a way to use an X100P as a trunk with DID numbers and all? We just bought one of these and want to create some VoIP extensions connected to our PBX as a trial. The PBX does not have capacity for any more T1 cards so it is the only cheap way for this trial. If not, what kind of hardware would you recommend to setup some analog extensions as DID trunks between a PBX and *? Not in the normal sense. But there is nothing stopping you from implementing a extension in asterisk like you would with DID and making the legacy PBX pickup the line and dial the number. -- 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
Re: [Asterisk-Users] Important: The Asterisk Mailing list(newsubject)
The only thing I hate more than not having a proper reply-to on a mailing list (one that replies to the LIST) is the people who havn't been on the net long enough to know how mailing lists work, and their whole function. Mailing lists are communities. The primary function is to share procedures, patches, fixes, workarounds, programming knowledge, etc.. with the rest of the community. Its the rare exception that once in a great while a topic strays off and goes personal/off-list. This should happen when the community cannot benefit from the discussion, such as a private deal for equipment (which is sometimes frowned upon with lists, but sometimes enjoyed), or some basic hand-holding that goes beyond the scope of the list, and that the rest of the listmembers should know. (I.E. someone asking how to setup Linux, thus not having anything to do with Asterisk, untill they get to the point where they can install Asterisk). So, my vote is to keep the reply-to as going to the list. Also, don't hijack subjects. If you are going to use reply insted of post, at least re-write the subject line! Please direct all flames privately, where they can be properly transfered to /dev/null - - - Jon Myers Online since 1985 (I know, not longer than alot of prople, but more than a couple years). At 07:30 PM 3/21/2004 -0600, you wrote: On Fri, 2004-03-19 at 22:23, Darren Nickerson wrote: Folks, I strongly support removing the current reply-to-list setting, and you should too. Like many new list admins, I once thought the reply-to was kewel. Requests to remove it kept coming up, ... usually around the same time someone embarrassed themselves by posting a personal reply/flame to the list. Someone, in frustration, finally pointed me to the following URL: http://www.unicom.com/pw/reply-to-harmful.html I saw the light. Please can the list admin step in and end this thread by either: a) announcing that the reply-to override has been removed b) announcing their resignation ;-) I'm sorry you saw the wrong light. You are peering into a light that will anger many more of us to the point of removing ourselves from the list as it becomes impossible to filter appropriately. Reply to group has a nasty habit of piling up addresses and then people who have dropped out of the thread are still getting barraged by messages where their address is still a part of it. It is bad enough we have users too lame to click on a link to the submission url and instead just reply and erase old content, your suggestion would just make people more likely to get nailed with unrelated content. Open source software thrives by efficient and open communications. To start suggesting people take useful commentary off list by making it less easy to reply to the list only reduces our resources. It also starts a lot of private communications and possibly private flame wars. If you post embarrassing information, or if your post embarrass you in public, maybe they didn't need to be said in the first place. Your aversion to fixing a to line when you take a message off list is not worth breaking good mail filtering. You can probably blame me for the original switch of the Reply-To header. I believe I am the one who requested it soo long ago. -- 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] Important: The Asterisk Mailing list(newsubject)
Strongly Agree :) WW - Original Message Follows - The only thing I hate more than not having a proper reply-to on a mailing list (one that replies to the LIST) is the people who havn't been on the net long enough to know how mailing lists work, and their whole function. Mailing lists are communities. The primary function is to share procedures, patches, fixes, workarounds, programming knowledge, etc.. with the rest of the community. Its the rare exception that once in a great while a topic strays off and goes personal/off-list. This should happen when the community cannot benefit from the discussion, such as a private deal for equipment (which is sometimes frowned upon with lists, but sometimes enjoyed), or some basic hand-holding that goes beyond the scope of the list, and that the rest of the listmembers should know. (I.E. someone asking how to setup Linux, thus not having anything to do with Asterisk, untill they get to the point where they can install Asterisk). So, my vote is to keep the reply-to as going to the list. Also, don't hijack subjects. If you are going to use reply insted of post, at least re-write the subject line! Please direct all flames privately, where they can be properly transfered to /dev/null - - - Jon Myers Online since 1985 (I know, not longer than alot of prople, but more than a couple years). At 07:30 PM 3/21/2004 -0600, you wrote: On Fri, 2004-03-19 at 22:23, Darren Nickerson wrote: Folks, I strongly support removing the current reply-to-list setting, and you should too. Like many new list admins, I once thought the reply-to was kewel. Requests to remove it kept coming up, ... usually around the same time someone embarrassed themselves by posting a personal reply/flame to the list. Someone, in frustration, finally pointed me to the following URL: http://www.unicom.com/pw/reply-to-harmful.html I saw the light. Please can the list admin step in and end this thread by either: a) announcing that the reply-to override has been removed b) announcing their resignation ;-) I'm sorry you saw the wrong light. You are peering into a light that will anger many more of us to the point of removing ourselves from the list as it becomes impossible to filter appropriately. Reply to group has a nasty habit of piling up addresses and then people who have dropped out of the thread are still getting barraged by messages where their address is still a part of it. It is bad enough we have users too lame to click on a link to the submission url and instead just reply and erase old content, your suggestion would just make people more likely to get nailed with unrelated content. Open source software thrives by efficient and open communications. To start suggesting people take useful commentary off list by making it less easy to reply to the list only reduces our resources. It also starts a lot of private communications and possibly private flame wars. If you post embarrassing information, or if your post embarrass you in public, maybe they didn't need to be said in the first place. Your aversion to fixing a to line when you take a message off list is not worth breaking good mail filtering. You can probably blame me for the original switch of the Reply-To header. I believe I am the one who requested it soo long ago. -- 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 Willy Wouters ypOne Publishing ___ 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] Cisco 7960 vs 7905
I stand corrected. I assume that the info at http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item09186a008010a826.shtml is referring to First Customer Ship, rather than current, since it lists the 7905G as not supporting SIP. And I KNOW the 7905G supports SIP. I was using one last week. --Eric On Sun, 2004-03-21 at 18:43, Walker Haddock wrote: On Sun, Mar 21, 2004 at 05:53:00PM -0600, Rich Adamson wrote: The 7940/7960 have been around for a long time while the 7905 is a rather recent addition to their product line. I believe the 7905 only supports the Cisco proprietary firmware (not sip) while the 7960 supports either Cisco or sip. That's probably why you're seeing v1 verses v6.3 or whatever. The 7905 and the 7905G both run the 1.01 SIP firmware. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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] AGI startup on channel when asterisk starts
On Sun, 2004-03-21 at 16:05, Jerry Geis wrote: All, I am looking for a way to have my AGI startup on a channel automatically when asterisk starts. Is this possible? I have my AGI working for when a call comes in - however I would like the AGI started up automatically with asterisk on a couple channels as I want to monitor my database and when things happen place a couple calls etc. I am aware of the outgoing directory but that is not exactly what I am wanting to do unless there is a way to have the commands in the file not actually dial but just give me a channel and start my AGI. They my AGI can place the call if that is what is required. As I told you one of the very few private mails I respond too, This is not what you want to do. AGI should not dial out as it will not continue to process the call at that point. Use AGI to handle the inbound calls, use a separate monitor app if you need to to initiate calls that will then be dropped in your AGI. Spend a little time learning and reading the list. you will find examples of each stage of your request. If you don't understand the need for the separation, continue reading the list until you do. Don't try to shoe horn asterisk into your idea of how things should work. Allow you mind to open to the way asterisk does things and you will eventually understand the extra flexibility available to you. -- 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] Home users
I am trying to setup the following scenario. 7960 --- Linksys firewall Internet Firewall Linux server 7960 Home Running Office Asterisk From the 7960 at my home I get connected. I can then call any other phone in the office and call outside calls. The problem is as soon as someone picks up their office phone there is dead silence. The office phone can call my home phone and it rings and again when I pick up the home phone there is nothing. I do have port forwarding turned on my office firewall. Can someone help me here? I am almost out of hair on my head. --Damian ___ 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] Important: The Asterisk Mailing list(newsubject)
On Sun, 2004-03-21 at 20:48, Jon Myers wrote: Online since 1985 (I know, not longer than alot of prople, but more than a couple years). But apparently not long enough to know that top posting and not trimming quotes are both just as bad as reply-to-sender. ;-) dk ___ 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] Cisco 7960 vs 7905
You're positive the 7905G supports SIP? How did you upgrade it? Just a TFTP server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Sunday, March 21, 2004 9:08 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Cisco 7960 vs 7905 I stand corrected. I assume that the info at http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0 9186a008010a826.shtml is referring to First Customer Ship, rather than current, since it lists the 7905G as not supporting SIP. And I KNOW the 7905G supports SIP. I was using one last week. --Eric On Sun, 2004-03-21 at 18:43, Walker Haddock wrote: On Sun, Mar 21, 2004 at 05:53:00PM -0600, Rich Adamson wrote: The 7940/7960 have been around for a long time while the 7905 is a rather recent addition to their product line. I believe the 7905 only supports the Cisco proprietary firmware (not sip) while the 7960 supports either Cisco or sip. That's probably why you're seeing v1 verses v6.3 or whatever. The 7905 and the 7905G both run the 1.01 SIP firmware. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 ___ 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] Cisco 7960 vs 7905
Hello, I own 2 7905G phones running sip. You can download the image from CCO. It is installed by TFTP, you just specify the server inside the menu, reboot the phone and if you have the image and the config file with the image in the root directory it will install the new OS. Regards, Matthew Enger [EMAIL PROTECTED] On Mon, 2004-03-22 at 13:25, Matthew Marlowe wrote: You're positive the 7905G supports SIP? How did you upgrade it? Just a TFTP server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Sunday, March 21, 2004 9:08 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Cisco 7960 vs 7905 I stand corrected. I assume that the info at http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0 9186a008010a826.shtml is referring to First Customer Ship, rather than current, since it lists the 7905G as not supporting SIP. And I KNOW the 7905G supports SIP. I was using one last week. --Eric On Sun, 2004-03-21 at 18:43, Walker Haddock wrote: On Sun, Mar 21, 2004 at 05:53:00PM -0600, Rich Adamson wrote: The 7940/7960 have been around for a long time while the 7905 is a rather recent addition to their product line. I believe the 7905 only supports the Cisco proprietary firmware (not sip) while the 7960 supports either Cisco or sip. That's probably why you're seeing v1 verses v6.3 or whatever. The 7905 and the 7905G both run the 1.01 SIP firmware. -- Matthew Enger [EMAIL PROTECTED] Mob: 0412 463 080 Direct: (03) 9747 4001 X Integration A Netcruiser Pty Ltd business Ph: 1300 730 997 Fax: 1300 136 720 ___ 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] Home users
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damian Dicks Sent: Sunday, March 21, 2004 9:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Home users I am trying to setup the following scenario. [...] there is nothing. I do have port forwarding turned on my office firewall. Can someone help me here? I am almost out of hair on my head. [...] canreinvite=no in sip.conf for the home phone. Without that, the phoen are trying to talk to each other directly, which isn't going to work when they are both (presumably) behind different NAT boxes. Canreinvite=no will force your home phone to always pass its traffic through the * box, eliminating the issue you are having. Daryl ___ 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] If you know your party's extension # please dial it now ...
If you have your IVR under context [mainmenu] and your extensions under
context [default]. Then make sure you include context default under
context mainmenu...
Because your mainmenu context does not know about any other extensions
if you don't.
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Posted At: Sunday, March 21, 2004 8:37 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] If you know your party's extension #
please dial it now ...
Subject: [Asterisk-Users] If you know your party's extension # please
dial it now ...
Hi all,
I've built the usual press one for sales, 2 for support IVR which
works fine but I'm having difficulty in allowing callers to type in
whole extension numbers.
My internal extn ranges are 3xxx and 4xxx. I have pasted the IVR below
(just in case someone wants one). The welcome message states callers
should type in the extension number they want or choose from the
options. It seems though that one can only press one number before the
IVR moves to the next step.
I'm starting to think that if my extn's are 3xxx and 4xxx I can't have
any menu choices beginning with 3 or 4. Would this be correct? If so how
does the received DTMF break out of the IVR and get matched to the
relevant dialplan entry?
[mainmenu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,5
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test exten
= s,5,Background(welcomemsg) exten = s,6,Background(choosemsg)
; Sales
exten = 1,1,Dial,SIP/3400|20
exten = 1,2,Voicemail(3400)
exten = 1,3,Goto(mainmenu,s,60
; Tech support
exten = 2,1,Dial,SIP/3401|20
exten = 2,2,Voicemail(3401)
exten = 2,2,Goto(mainmenu,s,1)
; Echo Test
exten = 3,1,Playback(demo-echotest)
exten = 3,2,Echo
exten = 3,3,Playback(demo-echodone)
exten = 3,4,Goto(mainmenu,s,6)
; Parrot Test
exten = 4,1,Goto(205,1)
; Access VoiceMail
exten = 5,1,VoicemailMain
exten = 5,2,Goto(mainmenu,s,6)
; Play the weasels
exten = 6,1,Wait,3
exten = 6,2,Playback(tt-somethingwrong)
exten = 6,3,Playback(tt-weasels)
exten = 6,4,Wait,2
exten = 6,5,Goto(mainmenu,s,6)
; # to hangup
exten = #,1,Playback(vm-goodbye)
exten = #,2,Hangup
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension #
exten = _3XXX,1,Dial(SIP/{EXTN}|20|T)
exten = _4XXX,1,Dial(SIP/{EXTN}|20|T)
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/ ___
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Re: [Asterisk-Users] Important: The Asterisk Mailinglist(newsubject)
At 09:34 PM 3/21/2004 -0500, you wrote: On Sun, 2004-03-21 at 20:48, Jon Myers wrote: Online since 1985 (I know, not longer than alot of prople, but more than a couple years). But apparently not long enough to know that top posting and not trimming quotes are both just as bad as reply-to-sender. touché I used to post in between lines, to respond to each point seperately, then people would say it confused them. Then I bottom posted, and people started saying that they didnt get my message, and instead got their message back with a bunch of arrows in front (not bothering to scroll down). So I've taken the approach of top posting, and allowing users to scroll down to see the origional post for reference. Everyone has their favorites, and preferences. Hard to come up with a standard, just like the 'ol reply-to thing. So if there are several points, I sometimes do alot of trimming, and respond within the quotes (double space after/before quote). If its a whole new line of thought, then a top post, if responding to a one-two liner (like this) then top quote trimmed, bottom post. Sometimes I think I put too much thought into things, and get caught up in the mess of trying to come up with a pseudo standard... (sigh) - - - Jon Myers ___ 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] LipZ4 Sip Soft Phone
Hi, From: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LipZ4 Sip Soft Phone Date: Sat, 20 Mar 2004 09:33:42 -0500 (EST) Thanks a lot I might give it a try. Any specific instructions for running it with asterisk? AJ Checkout these urls, these might be of your interest: http://www.zultys.com/products/lipz4/softphone-1.3.11-0.i386.rpm http://www.zultys.com/products/lipz4/lipz4_quick_start.pdf http://www.zultys.com/download_manuals.htm Regards, Girish _ Protect your PC from viruses. Get in the experts. http://www.msn.co.in/pcsafety/ Click here now! ___ 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
