Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
I am trying to locate the manual for that level software. If it's not here
at home it is at my office and I will look everything up in the morning.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, April 23, 2005 9:00 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P / F25.0
09-FEB1994 when I look up the software on the switch board so if I am
reading what your telling me then I have to do D4/AMI. So does my zaptel
look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Of course there are exceptions to the rules. I see now on a couple
software releases where they do allow PRI with D4/AMI and PRI with
esf/b8zs. It's been a year or so since I messed with trunking on a 200,
I've mostly been installing and maintaining the SX2000's and 3300's.
Henry
- Original Message -
From: Dennis Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and em channels
Here's parts of my config that takes the pri and converts it to em with
ANI DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and
this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and
modems
exten = 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten = _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00262.htmlFile: ATT00263.txt
I was wrong. I just looked in my Mitel IM's. What level software are
you
on in the SX200? Up until a certain level 200's could only do D4/AMI
T1's,
they could not do PRI's. If it is a newer switch within the past 3
years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.
Henry
- Original Message -
From: Scott Wolfe
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Friday, April 22, 2005 7:04 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks,
This is what I have now, but my Mitel PBX and Asterisk Box are unable
to communicate via the T1 connection. Asterisk loads ok but I get error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu
-Scott
/etc/zaptel.conf
loadzone = us
defaultzone=us
span=1,0,0,d4,ami
bchan=1-23
dchan=24
/etc/asterisk/zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=dms100
rxwink=300
usecallerid=no
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0 ;into the pstn twords the telco
txgain=0.0
callgroup=1
pickupgroup=1
immediate=yes
signalling=pri_cpe
group=1
context=default
emdigitwait=500
channel = 1-23 ; Set this to 1-15,17-31 for E1
- Original Message -
From: Michael D Schelin
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday,
Re: [Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
On 20:47, Sat 23 Apr 05, Stefan Gofferje wrote: Besides chan_capi does not understand Busy() and Congestion(), that probably is a matter of how fast the faxmodem picks up the call and to what value your timeout is set. However, I think, I have another strategy regarding security... I have separate contexts for incoming calls, and each outgoing line... I also have separate contexts for phones, one for fully trusted and one for remote clients. snip [default] exten = s,1,Hangup I also have exten = i,1,Goto(playrejectmessageandhangup,s,1) exten =ti,1,Goto(playrejectmessageandhangup,s,1) at the end of each incoming context, so it's rather unlikely that somebody ever even enters the default context... /snip Sounds like a setup better then mine. The wait always looked evil to me ;) Maybe I should set the faxmachine to pickup right away instead of ringing twice before picking up. The fax also has a phone in it, that's prolly why it is set to ring twice now. I took over admin here, so not my call to start with. Thanks for the clear setup information :) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
Has anybody success with speed dialing?
If so, I am sure you can help me to get into this club.
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was hoping to hear if it helped you to change that?
Let's try it together:
1. Open IPswitch
2. Open Extensions tab on top
3. Switch to the tab Speed Dials on the bottom
4. Fill in:
Name: [EMAIL PROTECTED]
Caller Id: Peter
Visible on Panel: (ticket)
Exentension Group: Speed Dial Numbers
Congratualtions, you have successfully installed the Asterisk Open
Source .
bye
Ronald
Thorben
Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
tgj wrote:
Hi Ronald,
I must admit I am getting confused now.
I understand that you have a problem getting Speed Dial Buttons to work.
The problem as I understand it is that the calls are placed in the wrong
context.
To solve that problem I have asked you to make sure that you have typed a
valid context on the configuration page. Have you tried that?
I think thats all you need to do, how do I post an example of that? It's a
fairly easy thing to do.
Thorben
What is the right syntax to do that?
Context for dialing a trunk line is trunkint
Peter has the phone number 011-234-5678
How to set it up as a speed dial number? Below are all info you may need:
The phone 601 (= Monitor extension) is a Sip phone,
[general]
context=default; Default context for incoming calls
[601]
type=friend
username=601
secret=dont+tell+you
canreinvite=no
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
nat=yes
callgroup=1
pickupgroup=1
callerid=Ronald Hotline,601
qualify=1000
extensions.conf
[default]
...
include = trunkint
...
[trunkint]
;
; International long distance through trunk
; . other lines deleted
exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,108,hangup
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[Asterisk-Users] help:Memory Consumption
Can anybody help me to figure out how much memory per minute is consumed for voicemail applications? And how many concurrent calls can be handled at a time in Asterisk? so that, I can choose the specification for Server to setup Asterisk for a large number of users. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy and Debian
Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Netjet/Linux/Asterisk issue
Hi All, Debian Sarge (most recent update yesterday). Running a custom-built 2.6.8 kernel (Debian kernel doesn't have the Traverse transparent mode patch - the 2.4 patch seemed to apply to the 2.6 sources OK). :01:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 0001:000e Flags: bus master, medium devsel, latency 220, IRQ 11 I/O ports at d400 [size=256] Memory at ff8fd000 (32-bit, non-prefetchable) [size=4K] *CLI show version Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 running Linux (Typing this, I wonder if the 'BRIstuffed' in the version number is relevant to my problem) I can dial out from a SIP phone fine - all works as expected. If I dial in via the BRI to a SIP phone, regardless of which end hangs up first, I get: Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success Apr 24 16:22:52 WARNING[2191]: chan_modem_i4l.c:391 i4l_read: Read failed: Success ...and so on. Once this happens the BRI channel is wedged until I restart Asterisk. Has anyone seen this behaviour? I Google'd for a good while but didn't turn anything up :-( I realise it could be the 2.4 Traverse patch not working with the 2.6 netjet.c code, or the 'BRIstuffed' Debian package of Asterisk. I'm hoping someone has seen this and can save me a few hours work :-) In the absence of any advice I'll be trying a 2.4 kernel tomorrow, time permitting. I suppose, beyond that, I'll try a build-from-sources version of Asterisk (although for this application I prefer the Debian package). Cheers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] ztdummy and Debian
Title: Nachricht - having the kernel sources or kernel headers installed - uncommenting ztdummy in zaptel'sMakefile - make / make install :) assuming you have an uhci chip on your main board and kernel 2.4x With kernel 2.6 make a make linux26 and things are more easy regards Manuel -Ursprüngliche Nachricht-Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von ThoreGesendet: Sonntag, 24. April 2005 13:15An: ASTERIKSBetreff: [Asterisk-Users] ztdummy and Debian Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy and Debian
Hi, This is how I got ztdummy on debian sarge: $ apt-get install kernel-headers-2.6.8-2-386 dpatch kernel-package zaptel zaptel-source $ cd /usr/src $ ln -s kernel-headers-2.6.8-2-386/ linux $ cd linux $ make-kpkg modules_image $ dpkg -i ../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb Selecting previously deselected package zaptel-modules-2.6.8-2-386. (Reading database ... 51551 files and directories currently installed.) Unpacking zaptel-modules-2.6.8-2-386 (from .../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb) ... Setting up zaptel-modules-2.6.8-2-386 (1.0.7-3+10.00.Custom) ... $ depmod -a $ modprobe ztdummy $ dmesg look at this (Les plus / Kit Zaptel) http://terminaux.levinux.org/wakka.php?wiki=LaTelephonie bye, samuel Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Samuel T. Cossette 1.418.8o2.784o ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QSIG.
Hi, Does * support QSIG?Some experience with it? Which card are adequated? Regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Announcement
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Re: [Asterisk-Users] ztdummy and Debian
Hi ! I this working with kernel 2.4? Thore - Original Message - From: Samuel T. Cossette [EMAIL PROTECTED] To: Thore [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 24, 2005 1:45 PM Subject: Re: [Asterisk-Users] ztdummy and Debian Hi, This is how I got ztdummy on debian sarge: $ apt-get install kernel-headers-2.6.8-2-386 dpatch kernel-package zaptel zaptel-source $ cd /usr/src $ ln -s kernel-headers-2.6.8-2-386/ linux $ cd linux $ make-kpkg modules_image $ dpkg -i ../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb Selecting previously deselected package zaptel-modules-2.6.8-2-386. (Reading database ... 51551 files and directories currently installed.) Unpacking zaptel-modules-2.6.8-2-386 (from .../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb) ... Setting up zaptel-modules-2.6.8-2-386 (1.0.7-3+10.00.Custom) ... $ depmod -a $ modprobe ztdummy $ dmesg look at this (Les plus / Kit Zaptel) http://terminaux.levinux.org/wakka.php?wiki=LaTelephonie bye, samuel Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Samuel T. Cossette 1.418.8o2.784o ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inband DTMF with IAX
Hi, I am currently having a problem where I am making outbound calls via IAX, these calls are then being routed by my provider through a SIP connection to a service providing PSTN access. The problem I have is the the DTMF is being sent inband over the SIP connection, and I am only receiving the DTMF inband on my IAX connection which asterisk is sensibly ignoring as from what I understand with IAX2 the DTMF should be sent out-of-band. I assume the simplest way to fix it is for my provider to put a dtmfmode=inband on the sip.conf entry for their PSTN provider, then presumably their asterisk would see the DTMF and send them out-of-band over the IAX2 channel, however, they are understandably wary of doing this in case it affects any of their other services which currently work perfectly... Is there any way to get Asterisk to listen for inband DTMF from an outbound IAX2 channel so that I can get round this problem in a simple way? (The reason I need to get DTMF on an outbound call is I am trying to set up a 'press 1 to accept the call' system for forwarding calls to mobiles). Thanks in advance, Alex Brett [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] ztdummy and Debian
Of course! The trick are the kernel headers but they must of course fit onto the installed kernel. The problem is: For the zaptel stuff you need more then the downloaded stuff. You need the kernel sources or headers. Manny -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Thore Gesendet: Sonntag, 24. April 2005 14:53 An: ASTERIKS Betreff: Re: [Asterisk-Users] ztdummy and Debian Hi ! I this working with kernel 2.4? Thore - Original Message - From: Samuel T. Cossette [EMAIL PROTECTED] To: Thore [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 24, 2005 1:45 PM Subject: Re: [Asterisk-Users] ztdummy and Debian Hi, This is how I got ztdummy on debian sarge: $ apt-get install kernel-headers-2.6.8-2-386 dpatch kernel-package zaptel zaptel-source $ cd /usr/src $ ln -s kernel-headers-2.6.8-2-386/ linux $ cd linux $ make-kpkg modules_image $ dpkg -i ../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb Selecting previously deselected package zaptel-modules-2.6.8-2-386. (Reading database ... 51551 files and directories currently installed.) Unpacking zaptel-modules-2.6.8-2-386 (from .../zaptel-modules-2.6.8-2-386_1.0.7-3+10.00.Custom_i386.deb) ... Setting up zaptel-modules-2.6.8-2-386 (1.0.7-3+10.00.Custom) ... $ depmod -a $ modprobe ztdummy $ dmesg look at this (Les plus / Kit Zaptel) http://terminaux.levinux.org/wakka.php?wiki=LaTelephonie bye, samuel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk2mp3
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[Asterisk-Users] VSAT and Asterisk
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[Asterisk-Users] Asterisk management GUI
Hi, I know it's a redundant question but this time it's different, I'm looking for a mature enough management and administration GUI to which I can further contribute and work with the developers to deliver a well featured package to meet with what people really demand, it's a shame that people would pay a lot for a cisco solution because it has a nice interface while they can pay less (or even nothing) if they can have the same performance and features, if not more, from an asterisk powered machine. What do you think? Ezabi signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk2mp3
Sorry, I forgot to show the script, here it is: #!/usr/bin/python import tempfile,os,sys,re,time monitordir = sys.argv[1] basename = sys.argv[2] def runcmd(cmd): print cmd os.system(cmd) #mix to one wav inwav = os.path.join(monitordir, basename+-in.wav) outwav = os.path.join(monitordir, basename+-out.wav) waste, mixedwav = tempfile.mkstemp(.wav,audiopipe_,/tmp) runcmd (soxmix %s %s %s % (inwav,outwav,mixedwav)) #up sample rate waste, uppedsamplerate = tempfile.mkstemp(.wav,audiopipe_,/tmp) runcmd(sox %s -r 22050 %s % (mixedwav, uppedsamplerate)) #run lame outfile = os.path.join(monitordir, basename+.mp3) #I use gogo instead of lame, cause gogo is a lot faster. But if you #can't compile gogo, lame will do just fine. # runcmd(lame -S -v %s %s % (uppedsamplerate, outfile)) #runcmd(/usr/local/bin/gogo -v 6 %s %s % (uppedsamplerate, outfile)) os.remove(inwav) os.remove(outwav) #but at least we can waste the temporaries os.remove(mixedwav) os.remove(uppedsamplerate) On 4/24/05, Stiffe [EMAIL PROTECTED] wrote: Hi folks. I have a problem with a pythonscript designed for joining one in-wav and one out-wav after recording a call. Yes, I have the wav-files after a successfull recording... Python stops at line 5:basename = sys.argv[2] like: [EMAIL PROTECTED] bin]# python /usr/local/bin/asterisk2mp3.py Traceback (most recent call last): File /usr/local/bin/asterisk2mp3.py, line 5, in ? basename = sys.argv[2] IndexError: list index out of range What do I have to do? And finally, can I have the script to mail the resulting mp3 to a mailadress and after that delete the mp3-file, for saving space? Thanks in advance //Stefan (from Sweden) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astcc Working but Can't Make The Call
Hello everyone: I have just reinstalled asterisk and astcc. Asterisk is working fine, but not astcc. When I try to make iax call through voipjet, Astcc is working fine till the pont where it tries to make the call, it gives a congestion message. Here is the message i get when i attempt to make the call through astcc: Called [EMAIL PROTECTED]/.. -- IAX2/64.28.107.46:4569-1 is circuit-busy Apr 24 05:13:41 NOTICE[30453]: chan_iax2.c:2761 auto_congest: Auto-congesting call due to slow response -- Hungup 'IAX2/64.28.107.46:4569-1' == Everyone is busy/congested at this time (1:0/1/0) -- AGI Script astcc.agi completed, returning 0 The ip address 64.28.107.46 is the voipjet ip, but not the one i defined in iax [voipjet] context. host= 216.118.117.46 This is the call records: cardnum | callerid | callednum | trunk | disposition 7799| unknown | 17046872001 | NULL | CONGESTION 7632| unknown | 17046872001 | NULL | CONGESTION 7632| unknown | 500 | NULL | CONGESTION Whem i make the call directly without astcc, the call goes through fine: dial -- Executing SetCallerID(OSS/dsp, ...) in new stack -- Executing Dial(OSS/dsp, IAX2/[EMAIL PROTECTED]/) in new stack -- Called [EMAIL PROTECTED]/ -- Call accepted by 216.118.117.46 (format gsm) -- Format for call is gsm -- IAX2/voipjet-1 is making progress passing it to OSS/dsp Any ideas please, i didn't get this message before i reinstalled astcc in an attempt fresh documentation Thanks; __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meetme Announcement
In article [EMAIL PROTECTED], Mohamed Farid [EMAIL PROTECTED] wrote: Dear All : How can I enable the announcement Feature of Meet-me rooms ? So that when I enter the conference room , the system ask me about my name ,, then announce all the existing people in the room about my entrance .. Also when I go out of the conference - an announce should be played to all the remaining members in the conference saying that I am out ... Pass the 'i' option to MeetMe in your dialplan. Note that this option is only available in CVS HEAD, not in STABLE nor in any of the 1.0.x releases. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registerport 5060 or 1720?
When do you use Registerport 5060 and when 1720 ?? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 EM wink issues - bad int'l dial-out and occasional dropped calls
Anyone have any ideas here? We are using 8 channels of EM Wink with a T100P for outgoing LD and incoming tollfree numbers and are apparently connected to a Nortel DMS-250 at the CO. We are receiving ANI DNIS just fine and can dial-out domestically with DTMF but have two issues that are still unresolved: 1) We cannot dial-out internationally with an 011 prefix (or any other prefix that we can think of). Qwest claims (1) they never get international calls and (2) domestic calls are routed to their LD service as 1NXXNXX instead of 1NXXNXX. Is some form of prefix/suffix needed for DTMF dialing over an EM wink channel? (e.g. something like the *ANI*DNIS* for incoming.) 011 clearly doesn't work as a prefix and Qwest's response has invariably been 'there is something wrong with your PBX' :( Curiously if we follow an 011+international-number with a * we get a recording that we have not entered sufficient digits to complete the call whereas without the * we just get a congestion beep from the far end. 2) Once or twice a day the customer is getting calls dropped. The log shows the following: Apr 21 13:15:51 VERBOSE[22664]: -- Hungup 'Zap/7-1' Apr 21 13:15:52 VERBOSE[22664]: -- Starting simple switch on 'Zap/7-1' Apr 21 13:15:53 WARNING[22664]: getdtmf on channel 7: Operation now in progress Apr 21 13:15:53 VERBOSE[22664]: -- Hungup 'Zap/7-1' It appears that we see the line go back on-hook, hangup but then see it go off-hook again and treat it as another incoming call that never gets a DTMF input when in fact the call has just been dropped. We've verified that we are not sharing interrupts, we are on run level 3 etc. zttest shows (so far) a minimum of 99.987%. Can anyone think of what might be causing this or what we could ask Qwest regarding possible diagnostics? 3) Finally, what level of dropped calls is generally considered acceptable? Like the dead-pixel issue with LCDs this is pretty subjective but is there an industry number that is typical? (We are presently at ~1% due to this issue.) Thanks to all for any shared wisdom. Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QOS Routers
I run m0n0wall (http://m0n0.ch/wall) on a Soekris 4501 embedded PC (http://www.soekris.com). Very tweakable. Under $200. Michael On Fri, 22 Apr 2005 10:42:20 -0700, Max Clark wrote: Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks Henry,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 11:05 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
I am trying to locate the manual for that level software. If it's not here
at home it is at my office and I will look everything up in the morning.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, April 23, 2005 9:00 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P /
F25.0 09-FEB1994 when I look up the software on the switch board so if I
am reading what your telling me then I have to do D4/AMI. So does my
zaptel look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Of course there are exceptions to the rules. I see now on a couple
software releases where they do allow PRI with D4/AMI and PRI with
esf/b8zs. It's been a year or so since I messed with trunking on a 200,
I've mostly been installing and maintaining the SX2000's and 3300's.
Henry
- Original Message -
From: Dennis Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and em channels
Here's parts of my config that takes the pri and converts it to em
with
ANI DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS
through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to
dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and
this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and
modems
exten = 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten = _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00262.htmlFile: ATT00263.txt
I was wrong. I just looked in my Mitel IM's. What level software are
you
on in the SX200? Up until a certain level 200's could only do D4/AMI
T1's,
they could not do PRI's. If it is a newer switch within the past 3
years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.
Henry
- Original Message -
From: Scott Wolfe
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Friday, April 22, 2005 7:04 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks,
This is what I have now, but my Mitel PBX and Asterisk Box are
unable
to communicate via the T1 connection. Asterisk loads ok but I get error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu
-Scott
/etc/zaptel.conf
loadzone = us
defaultzone=us
span=1,0,0,d4,ami
bchan=1-23
dchan=24
/etc/asterisk/zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=dms100
rxwink=300
usecallerid=no
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0 ;into the pstn twords the telco
txgain=0.0
Re: [Asterisk-Users] Re: QOS Routers
This link points to a page about a switch...not a router. Michael On Fri, 22 Apr 2005 18:14:05 -0400, Iassen Hristov wrote: Maybe this fits the bill. http://www.gigafast.com/products/product_detail/EE2400-SS.htm It retails for less than $100 Message: 9 Date: Fri, 22 Apr 2005 10:42:20 -0700 From: Max Clark [EMAIL PROTECTED] Subject: [Asterisk-Users] QOS Routers To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feedback on Junction Networks conferences?
Hello everyone, I'm been toying with the idea of allowing my users to use meetme but have had some service quality issues (which I know are being addressed) but am concerned about making work for myself for something I can outsource... Junction Networks (http://www.junctionnetworks.com) seems to offer a pretty good booking and conferencing system for free if you connect via IAX/SIP - the ability to connect via the pstn is a good backup for my users as well. I assume since this is their primary business that they are running the patches and have the quality issues resolved - so I ask: Has anyone used this via iax or sip for several users and have any feedback they can share? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registerport 5060 or 1720?
The 5060 is usually SIP Proxy listen port. And the 1720 is usually h323 gatekeeper's listen port. On 4/24/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: When do you use Registerport 5060 and when 1720 ?? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cidsignailling mode question
Hi, I need to use cidsignalling=dtmf where the callerid comes after the first ring. Looking in source code of chan_zap.c I understand that cidsignalling=dtmf only works when cidstart=polarity. Is this right? or also works with cidstart=ring? Thanks Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can several Asterisk boxes working together?
The idea: 1. IPv6 is experimental, I would like to set it up as an extra box 2. MeetMe could kill my bandwidth. I would like to co-locate it. How can I combine different boxes / at different location to one system? How can I reach each other? Does anybody have experience in doing that? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTFM tones almost completly muted.
[EMAIL PROTECTED] wrote: On Fri, 22 Apr 2005, Peter Bowyer wrote: On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote: Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous DTMF tone of about 3 seconds) on the asterisk server (stored in the voice mail) the tone is more or less completely muted, just the initial tone start can be heard. I am using the G711 codec. Does anyone have any idea if these tones are on purpose muted by the service providers or any other reason why it does not work? Most likely the DTMF tones have been detected at the point where the call was converted PSTN-SIP/IAX, and forwarded instead as an indication (ie via SIP INFO or RFC2833 or whatever. So you won't hear them in a recording of the audio stream. The remaining blip is just the little bit at the start before the gateway recognised the tone. You should receive the indication in your SIP or IAX connection and Asterisk should see it (but its not audio any more). Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Steve, Good point, it makes sense that this is what is happening and most likely at the PSTN termination point. The question is where has the signalling gone as I seem not to receive it at my asterisk server. Do you think that this is a configuration problem at the PSTN terminators site or do they do this on purpose so they can charge extra for the information etc? Thanks. Ian Hailey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTFM tones almost completly muted.
Ian Hailey wrote: [EMAIL PROTECTED] wrote: On Fri, 22 Apr 2005, Peter Bowyer wrote: On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote: Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous DTMF tone of about 3 seconds) on the asterisk server (stored in the voice mail) the tone is more or less completely muted, just the initial tone start can be heard. I am using the G711 codec. Does anyone have any idea if these tones are on purpose muted by the service providers or any other reason why it does not work? Most likely the DTMF tones have been detected at the point where the call was converted PSTN-SIP/IAX, and forwarded instead as an indication (ie via SIP INFO or RFC2833 or whatever. So you won't hear them in a recording of the audio stream. The remaining blip is just the little bit at the start before the gateway recognised the tone. You should receive the indication in your SIP or IAX connection and Asterisk should see it (but its not audio any more). Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Steve, Good point, it makes sense that this is what is happening and most likely at the PSTN termination point. The question is where has the signalling gone as I seem not to receive it at my asterisk server. Do you think that this is a configuration problem at the PSTN terminators site or do they do this on purpose so they can charge extra for the information etc? Thanks. Ian Hailey. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users OK I found that it does work correctly with PSTN-IAX termination from voipuser.co.uk for example so it is realy a problem with sipgate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
Please contact me Urgent... Atentamente, Franz Schuverer Arrue GLOBAL GROUP, INC. www.telefoniaglobal.net [EMAIL PROTECTED] Tel. (504) 221-4062 (Honduras Tel. (507) 322-2259 (Panamá) Tel. (866) 978-0976 (U.S.A.) (SKYPE) franz1969 (MSN MESSENGER) [EMAIL PROTECTED] (YAHOO MESSENGER) [EMAIL PROTECTED] CONFIDENCIALIDAD. El contenido de esta comunicación, así como el de toda la documentación anexa, es confidencial y va dirigido únicamente al destinatario del mismo. En el supuesto de que usted no fuera el destinatario, le solicitamos que nos lo indique y no comunique su contenido a terceros, procediendo a su destrucción. CONFIDENCIALITY. The content of this communication and any attached information is confidential and exclusively for the use of the addressee. If you are not the addressee, we ask you to notify to the sender and do not pass its content to another person, and please be sure you destroy it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215
Hi all, What is the best client's protocol for my softphones in Windows pcs? and what is the best way for connecting clients, I meant should I use one protocol for all the clients or some mix (SIP/IAX/oh323) of protocols? what is better? Thanks in advance Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fritz+chan_misdn - any working example ?
Hi, I'd kindly ask if anyone can provide working configuration examples for Asterisk-Fritz-mISDN combo. Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the best client's protocol for my softphones
Hi all, What is the best client's protocol for my softphones in Windows pcs? and what is the best way for connecting clients, I meant should I use one protocol for all the clients or some mix (SIP/IAX/oh323) of protocols? what is better? Thanks in advance Kumara Sorry, I could not change the sub in previous one. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215
For my two cents .. IAX/IAX2 is the only way to go .. It stops most if not all Firewall issues as well as double NAT ... BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Sunday, April 24, 2005 10:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215 Hi all, What is the best client's protocol for my softphones in Windows pcs? and what is the best way for connecting clients, I meant should I use one protocol for all the clients or some mix (SIP/IAX/oh323) of protocols? what is better? Thanks in advance Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. On Apr 23, 2005, at 11:56 PM, Jaime Blanco wrote: Jerry, when you say digitmap, you mean in my extensions.conf file? Thanks. Jaime From: Jerry [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Date: Sat, 23 Apr 2005 19:44:20 -0500 Try adding a comma to your digitmap where you wish the dialtone to come back on. Works on a Polycom. On Apr 23, 2005, at 7:12 PM, Eric Wieling aka ManxPower wrote: Grandstream does not support a dialplan. It is supposed to support Early Dial, but didn't work. I've been told that recent firmware fixes the early dial bug. I doubt that Early Dial is the solution. The solution is to buy a good IP Phone. Polycom and SIPura both support continue dialtone after digit. Cisco ATAs do not. I don't know if the Cisco IP phones do or not. Alexander Lopez wrote: ignorepat is for Zapata devices. Sip devices sned the number to the swith AFTER the SIP device feels it has dialed it. I am not a pro on the GS phones, (never played with them) but I would cheak the documentation on setting up a 'dialplan'. I hope this sets you in the right direction. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaime Blanco Sent: Saturday, April 23, 2005 4:38 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ignorepat doesn't work Hi, I was trying to get the solution for the issue with getting dial tone after dialing 9, in sip phone, but I couldn't get anything. I am using a Grandstream Budgetone 100. I include ignorepat in the handset context, but nothing. Any guideline or help? Thanks. Jaime On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that ignorepat only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the ignorepat in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sm bounty validate length of e164/e212 number for all countries
I am building a simple form validation that needs to do a simple validation on the _length_ of a phone number. As we all know, different countries have different phone number lengths. For example, Australia phone numbers can be either 6 or 7 digits, while USA phone numbers are always 10 digits. I need a *CSV or database* of any kind that will simply give me the min and max for the phone number length. I need current data, but it does not have to be up to the minute. My form validation is _not_ going to be strict. It will just say Your country of origin is Australia. You only entered 5 digits, but Australian phone numbers are usually 6 or 7 digits. Do you want to proceed? If you can help me get this data, I will provide a small bounty for your help. I really appreciate it. The data I need is available at http://www.numberingplans.com/index.php?goto=download but it is way more data then I need, and $5000 per month. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
Jerry wrote: The digitmap is in your telephone. Used to terminate dialing and send the dialed string to *. Grandstream BT phones don't have a digitmap feature. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best client's protocol for my softphones
What is the best client's protocol for my softphones in Windows pcs? and what is the best way for connecting clients, I meant should I use one protocol for all the clients or some mix (SIP/IAX/oh323) of protocols? what is better? I would say IAX. If you only use 1 protocol for your clients, they can native-bridge, so Asterisk won't stay in the path. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to prevent native bridging between SIP channels
Hello, how can I prevent Asterisk from trying to create a native bridge between an incoming call from a SIP provider and an extension attached to a SIP ATA? My Asterisk is behind a firewall, and the native bridge invariably fails. Thanks in advance for any suggestion! (I DID search the list archives for native bridge and found one similar query without any replies). Regards, Wolf Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to prevent native bridging between SIP channels
add canreinvite=no to the sip user definition blocks for the SIP provider and for the SIP ATA. Regards, Marc Wolf N. Paul wrote: Hello, how can I prevent Asterisk from trying to create a native bridge between an incoming call from a SIP provider and an extension attached to a SIP ATA? My Asterisk is behind a firewall, and the native bridge invariably fails. Thanks in advance for any suggestion! (I DID search the list archives for native bridge and found one similar query without any replies). Regards, Wolf Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astcc Working but Can't Make The Call
I always answer myself because my mind would be tired at the time of asking the question. This time, i made a mistake in the trunk configuration in astcc web interface. peer/trunk was [EMAIL PROTECTED] and should've been [EMAIL PROTECTED] --- chawki hammoud [EMAIL PROTECTED] wrote: Hello everyone: I have just reinstalled asterisk and astcc. Asterisk is working fine, but not astcc. When I try to make iax call through voipjet, Astcc is working fine till the pont where it tries to make the call, it gives a congestion message. Here is the message i get when i attempt to make the call through astcc: Called [EMAIL PROTECTED]/.. -- IAX2/64.28.107.46:4569-1 is circuit-busy Apr 24 05:13:41 NOTICE[30453]: chan_iax2.c:2761 auto_congest: Auto-congesting call due to slow response -- Hungup 'IAX2/64.28.107.46:4569-1' == Everyone is busy/congested at this time (1:0/1/0) -- AGI Script astcc.agi completed, returning 0 The ip address 64.28.107.46 is the voipjet ip, but not the one i defined in iax [voipjet] context. host= 216.118.117.46 This is the call records: cardnum | callerid | callednum | trunk | disposition 7799| unknown | 17046872001 | NULL | CONGESTION 7632| unknown | 17046872001 | NULL | CONGESTION 7632| unknown | 500 | NULL | CONGESTION Whem i make the call directly without astcc, the call goes through fine: dial -- Executing SetCallerID(OSS/dsp, ...) in new stack -- Executing Dial(OSS/dsp, IAX2/[EMAIL PROTECTED]/) in new stack -- Called [EMAIL PROTECTED]/ -- Call accepted by 216.118.117.46 (format gsm) -- Format for call is gsm -- IAX2/voipjet-1 is making progress passing it to OSS/dsp Any ideas please, i didn't get this message before i reinstalled astcc in an attempt fresh documentation Thanks; __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quantum A800 (SIP) - Asterisk Config
Hi Is there any help for me to register my quantium A800 (SIP) with my Asterisk . Please help me what should me my Sip.conf now present i did [1234567] type=friend context=sip username= secret= nat=yes host=dynamic canreinvite=no defaultip=XXX.XXX.XXX.XXX disallow=all allow=g729 allow=gsm allow=g723.1 allow=ulaw and is there any special change need on quintum? Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best of the best of IP Phones
On Fri, 22 Apr 2005, Chris Coulthurst wrote: Is there a specific SIP or IAX phone that truly shines above the rest where it comes to 'happy' compatibility with Asterisk? I guess I'm talking about feature sets, like early-dial, off hook call announcing, conferencing, echo suppression, etc etc.. I, like many others, bought a Budgetone for early testing, and need some new eye candy! OHCA is a feature that I'd love to integrate, and it seems that not too many phones support it out of the box. I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom Soundpoint IP-500 and 600 to my Cisco's now. All things being equal between the phones, the following are why I prefer the Polycoms: 1. Better speakerphone than the Cisco 7960s. Despite the fact that Cisco licensed Polycom's SpeakerPhone technology, the SoundPoint IP 500 and 600 just sound and work better. 2. Lower price point: $185 for a NEW SoundPoint IP 500 is better than the $225 I see for used 7960s. 3. FTP based provisioning. TFTP is fine, but doesn't work very well through some NAT implementations. The PolyCom's can be centrally provisioned from any FTP server, and NAT doesn't seem to be a problem for it. 4. More intuitive User Interface. My clients require less training and are up and running quicker on the Polycoms. These are my opinions I love BOTH phones, and you can't go wrong with either choice, but for my needs the Polcom's work a lot better. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Asterisk do the following for me ?
Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need info on necessary config of new T1/PRIs
I have a T1 (PRIs) getting installed tomorrow and plan to plug it into a Sangoma A101. My question is are there any specifics I need to tell the CLEC's engineer regarding the configuration for Asterisk to see it? This is obviously new to me so any help is most appreciated. Regards, Jess ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - FIXED :-)
I finally figured it out ... working with BT100 you need to make a little voodoo ritual first :-) ... so follow the steps --exactly-- if you have trouble This is my working configuration behind Linksys WRT54G router: - Upgrade firmware 1.0.5.23 - Reset BT100 to factory defaults - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - DTMF: SIP INFO - Reboot BTW ... this is exactly what I tried 100x before but without the exact order of steps. I think especially step #2 about resetting to factory defaults before you do any re-configuration is critical. Don't trust the web interface always start fresh. Strangely, I had no problems whenever I was behind any other router than Linksys ... didn't have to do all this voodoo stuff ... makes me uncomfortable since I feel like I'll plug the phones in tomorrow and I'll be back where I started. Maybe the secret was not changing my underwear in the morning :-) LOL On the Asterisk side it's just the usual: Nat = yes Qualify = yes Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G - URI? I think I'm getting closer to figuring this out ... I just tried Linksys PAP2 and it registered just fine. I looked at the SIP packets captured by ethereal and I discovered that the real problem will probably be the uri in the authorization. For the working Linksys PAP2 and X-Lite I get: Authorization: DIGEST ... uri=sip:asterisk.mydomain.com ... For the BT100 which doesn't register (403 Forbidden) I get: Authorization: DIGEST ... uri=sip:wan-ip-of-the-router ... ... this kind of makes sense ... that looks like the wrong uri to send. So for some reason BT100 sends the wrong URI ... how can I fix this?? Again the weird thing is that if I plug in the BT100 behind any other router then Linksys WRT54G everything works fine. I'm trying my BT100 with the following config: - SIP Server: asterisk.mydomain.com - Outgoing Proxy: asterisk.mydomain.com - Nat travelsal: no - Local sip port: 5060 - Use NAT ip: no - Proxy require: no And in my sip.conf I have Nat=yes Qualify=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian Sent: Saturday, April 23, 2005 11:04 PM To: 'Pedro'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SIP registration behind Linksys WRT54G Yes that's the first thing I tried ... I'm able to make it work (using different routers than Linksys) in the following ways: - Set outgoing proxy and no STUN OR - No outgoing proxy and set STUN But once I put it behind Linksys everything registration does not work any more. Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Saturday, April 23, 2005 10:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Have you tried to enable NAT translation on the Grandstream? On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote: I'm trying to register BT100s ... (doesn't work) X-Lite seems to work though Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo Sent: Saturday, April 23, 2005 8:48 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G Oh yeah, duh.. Forgot.. I also have an SPA-2000 and a Cisco ATA-186 running behind my Linksys WTR43GS with no issues. This is at home registering to an external * box and to vonage. - Original Message - From: Luki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, April 23, 2005 9:41 PM Subject: Re: [Asterisk-Users] SIP registration behind Linksys WRT54G The WRT54G work fine... I have a Sipura 1000 and a Grandstream 286, both nated through a WRT54G on a single public IP. Worked out of the box -- no special settings needed. I was even surprised that I did not need to turn on the NAT handling in the Sipura ATA. Then I have a WRT54G running as a wireless client, and a Sipura 1001 connected to it, essentially behind two NAT's. Works fine too. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] T1 EM wink issues - bad int'l dial-out andoccasional dropped calls
Customer has integrated access arrangement with 16 channels of data/8 for voice that is split via customer cisco equipment. No local dialing, LD and incoming 800 service only via the t1. Qwest provides both the local loop and LD/800 service but it is provided via re-seller PNG. We have verified clock integrity via cisco logs that show no frame slips (cisco uses CO as a reference and we use the cisco as a reference.) No reference to Feature Group A (or D for that matter) is on our paperwork. It is nearly a rural location so I'm guessing we are connected via an End Office but can check this. Bill jltaylor wrote: What kind of service did you subscribe to (what do they call it on your bill)? Retail business trunks? Feature Group A? Can you dial local numbers or is this all long distance? Is Qwest the LEC or a long distance provider for this service? Are you connected to an End Office or a Tandem? These all may give me a hit as to what is going on. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of bill black Sent: Sunday, April 24, 2005 10:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 EM wink issues - bad int'l dial-out andoccasional dropped calls Anyone have any ideas here? We are using 8 channels of EM Wink with a T100P for outgoing LD and incoming tollfree numbers and are apparently connected to a Nortel DMS-250 at the CO. We are receiving ANI DNIS just fine and can dial-out domestically with DTMF but have two issues that are still unresolved: 1) We cannot dial-out internationally with an 011 prefix (or any other prefix that we can think of). Qwest claims (1) they never get international calls and (2) domestic calls are routed to their LD service as 1NXXNXX instead of 1NXXNXX. Is some form of prefix/suffix needed for DTMF dialing over an EM wink channel? (e.g. something like the *ANI*DNIS* for incoming.) 011 clearly doesn't work as a prefix and Qwest's response has invariably been 'there is something wrong with your PBX' :( Curiously if we follow an 011+international-number with a * we get a recording that we have not entered sufficient digits to complete the call whereas without the * we just get a congestion beep from the far end. 2) Once or twice a day the customer is getting calls dropped. The log shows the following: Apr 21 13:15:51 VERBOSE[22664]: -- Hungup 'Zap/7-1' Apr 21 13:15:52 VERBOSE[22664]: -- Starting simple switch on 'Zap/7-1' Apr 21 13:15:53 WARNING[22664]: getdtmf on channel 7: Operation now in progress Apr 21 13:15:53 VERBOSE[22664]: -- Hungup 'Zap/7-1' It appears that we see the line go back on-hook, hangup but then see it go off-hook again and treat it as another incoming call that never gets a DTMF input when in fact the call has just been dropped. We've verified that we are not sharing interrupts, we are on run level 3 etc. zttest shows (so far) a minimum of 99.987%. Can anyone think of what might be causing this or what we could ask Qwest regarding possible diagnostics? 3) Finally, what level of dropped calls is generally considered acceptable? Like the dead-pixel issue with LCDs this is pretty subjective but is there an industry number that is typical? (We are presently at ~1% due to this issue.) Thanks to all for any shared wisdom. Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Which protocol? was Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 9, Issue 215
On 24 Apr 2005, at 18:53, Kumara Jayaweera wrote: Hi all, What is the best client's protocol for my softphones in Windows pcs? and what is the best way for connecting clients, I meant should I use one protocol for all the clients or some mix (SIP/IAX/oh323) of protocols? what is better? Tricky question. Best to look at it the other way around. Which softphones can you use/license/buy? What features do you need? I use SIP for PCs on the office LAN because it gives me a wide choice of softphones and the internal firewalling is in my control. I use IAX over the WAN because it goes through NAT and firewalls much more easily, but there are fewer softphones. I never use H323. Tim http://www.westhawk.co.uk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Can Asterisk do the following for me ?
Why don't you use Vonage (what ever that might be :) to forward to a free account at a sip or iax phone provider somewhere in the world, make your European asterisk register with that account and dial out locally? :) Of course this and much furtehr similar works! :) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von [EMAIL PROTECTED] Gesendet: Sonntag, 24. April 2005 21:34 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
Excellent news! Now, remember that I am in Europe and out of my Linksys VoIP router I have a phone line coming out which I believe is North American standard. I am really not knowledgeable enough about the differences in both Networks so do I need a special card for Europe ? Otherwise, I'm set to purchase 2 X100P cards! The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
Don't forget you may like to support digium by buying an official tdm400P I know more expensive then a $7 clone but will work better on lines different to the 600ohm US pstn Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 4:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Excellent news! Now, remember that I am in Europe and out of my Linksys VoIP router I have a phone line coming out which I believe is North American standard. I am really not knowledgeable enough about the differences in both Networks so do I need a special card for Europe ? Otherwise, I'm set to purchase 2 X100P cards! The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 EM wink issues - bad int'l dial-out and occasional dropped calls
bill black wrote: Anyone have any ideas here? We are using 8 channels of EM Wink with a T100P for outgoing LD and incoming tollfree numbers and are apparently connected to a Nortel DMS-250 at the CO. We are receiving ANI DNIS just fine and can dial-out domestically with DTMF but have two issues that are still unresolved: 1) We cannot dial-out internationally with an 011 prefix (or any other prefix that we can think of). Qwest claims (1) they never get international calls and (2) domestic calls are routed to their LD service as 1NXXNXX instead of 1NXXNXX. Is some form of prefix/suffix needed for DTMF dialing over an EM wink channel? (e.g. something like the *ANI*DNIS* for incoming.) 011 clearly doesn't work as a prefix and Qwest's response has invariably been 'there is something wrong with your PBX' :( Curiously if we follow an 011+international-number with a * we get a recording that we have not entered sufficient digits to complete the call whereas without the * we just get a congestion beep from the far end. 2) Once or twice a day the customer is getting calls dropped. The log shows the following: Apr 21 13:15:51 VERBOSE[22664]: -- Hungup 'Zap/7-1' Apr 21 13:15:52 VERBOSE[22664]: -- Starting simple switch on 'Zap/7-1' Apr 21 13:15:53 WARNING[22664]: getdtmf on channel 7: Operation now in progress Apr 21 13:15:53 VERBOSE[22664]: -- Hungup 'Zap/7-1' It appears that we see the line go back on-hook, hangup but then see it go off-hook again and treat it as another incoming call that never gets a DTMF input when in fact the call has just been dropped. We've verified that we are not sharing interrupts, we are on run level 3 etc. zttest shows (so far) a minimum of 99.987%. Can anyone think of what might be causing this or what we could ask Qwest regarding possible diagnostics? 3) Finally, what level of dropped calls is generally considered acceptable? Like the dead-pixel issue with LCDs this is pretty subjective but is there an industry number that is typical? (We are presently at ~1% due to this issue.) Thanks to all for any shared wisdom. Bill http://www.qwest.com/largebusiness/products/voice/callingcards/lb_dial_guide.html based on that info, i'd say you are about to have a very crappy day. G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF intermittently stops working
Joseph wrote: We have the same problem with 7960, just randomly it will stop *hearing* the dtmf tones and you have to hangup and call back. This problem was fixed in CVS long ago, and current stable releases have the fix as well. When you are running a copy of Asterisk that is 4/5 months old, it's better to update first before reporting a problem, since it may already have been fixed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
According to the Mitel manuals that version of SX-200D can only use a
regular 24 channel T1. It can not use a PRI interface. You are going to
have to configure * to use a standard T1 not a PRI D4/AMI is the correct
signaling.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 11:09 AM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks Henry,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 11:05 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
I am trying to locate the manual for that level software. If it's not
here at home it is at my office and I will look everything up in the
morning.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Sent: Saturday, April 23, 2005 9:00 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
The Switch is since 1995 and I get a SX-200 Digital G1005 ENH 672P /
F25.0 09-FEB1994 when I look up the software on the switch board so if I
am reading what your telling me then I have to do D4/AMI. So does my
zaptel look correct? Maybe my cableing is off.
Thanks,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 8:34 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Of course there are exceptions to the rules. I see now on a couple
software releases where they do allow PRI with D4/AMI and PRI with
esf/b8zs. It's been a year or so since I messed with trunking on a
200, I've mostly been installing and maintaining the SX2000's and
3300's.
Henry
- Original Message -
From: Dennis Walker [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 9:13 PM
Subject: RE: [Asterisk-Users] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
My sx200 can only do ami d4 and em channels
Here's parts of my config that takes the pri and converts it to em
with
ANI DNIS
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS
through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to
dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and
this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and
modems
exten = 1234,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = ,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 5000 - 5199 to the SX200
exten = _5[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to
dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 8:56 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00262.htmlFile: ATT00263.txt
I was wrong. I just looked in my Mitel IM's. What level software
are you
on in the SX200? Up until a certain level 200's could only do D4/AMI
T1's,
they could not do PRI's. If it is a newer switch within the past 3
years
or an older switch with later software than you can do PRI, but the
signaling and framing must be ESF/B8ZS.
Henry
- Original Message -
From: Scott Wolfe
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Friday, April 22, 2005 7:04 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
Thanks,
This is what I have now, but my Mitel PBX and Asterisk Box are
unable
to communicate via the T1 connection. Asterisk loads ok but I get
error
lights (blinking orange) on my TE110P and on my Mitel T1 card. Hu
[Asterisk-Users] g729 passthrough?
I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Format for call is g729 Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . . I get ringback from Nufone, but as soon as the call answers I get an error: Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . What am I doing wrong to cause it to want to transcode? I assume that's where the complaint is coming from. I thought Asterisk could pass through without transcoding as long as the endpoints are all g729. Thanks. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 EM wink issues - bad int'l dial-out and occasional dropped calls
http://www.qwest.com/largebusiness/products/voice/callingcards/lb_dial_guide.html based on that info, i'd say you are about to have a very crappy day. G sorry to reply to my own post, forgot to suggest trying to send calls over another network. http://www.thedigest.com/faq/picodes.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best of the best of IP Phones
Greg Boehnlein wrote: I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom Soundpoint IP-500 and 600 to my Cisco's now. All things being equal between the phones, the following are why I prefer the Polycoms: 1. Better speakerphone than the Cisco 7960s. Despite the fact that Cisco licensed Polycom's SpeakerPhone technology, the SoundPoint IP 500 and 600 just sound and work better. 2. Lower price point: $185 for a NEW SoundPoint IP 500 is better than the $225 I see for used 7960s. 3. FTP based provisioning. TFTP is fine, but doesn't work very well through some NAT implementations. The PolyCom's can be centrally provisioned from any FTP server, and NAT doesn't seem to be a problem for it. 4. More intuitive User Interface. My clients require less training and are up and running quicker on the Polycoms. These are my opinions I love BOTH phones, and you can't go wrong with either choice, but for my needs the Polcom's work a lot better. The Polycoms also include a power supply and SIP firmware, which the Ciscos do not. Overall I just think the Polycoms are a better value. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 passthrough?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch Sent: Sunday, April 24, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] g729 passthrough? I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Format for call is g729 Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . . I get ringback from Nufone, but as soon as the call answers I get an error: Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . What am I doing wrong to cause it to want to transcode? I assume that's where the complaint is coming from. I thought Asterisk could pass through without transcoding as long as the endpoints are all g729. Thanks. B. ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. James Taylor MetroTel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
jltaylor wrote: ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. Still no joy. Added the allow=g729 to general, too, and I still get the same errors. Thanks anyways. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 EM false busy after dial
TE101P card T1 EM trunk to telco
on a SIP-PSTN call, after dial
SIP phone hears two seconds busy tone (1) then ring tone
how do we get rid of busy tone?
(1) two second busy
(480+620/500 0/500 480+620/500 0/500)
---
extensions.conf:
;
; dial-out to the PSTN with 7 digits
;
exten = _NXX,1,Dial(Zap/g1/${EXTEN})
exten = _NXX,n,Hangup()
zaptel.conf:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
zapata.conf:
[trunkgroups]
[channels]
language=en
context=default
signalling=featb
usecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=8
channel = 1-24
--
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RE: [Asterisk-Users] T1 EM false busy after dial
If Feature Group B signaling is working properly (and you have Feature Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is 1
or 0 based on the number assigned to you}.
If you are dialing out {terminating where you look like the carrier} on
FGB then it depends on if you are connected to an Equal Access End Office or
a Access Tandem.
Are you sure about the Feature Group B thing or do you have trunks that just
require MF signaling?
If you want MF, you might try the featdmf setting, however, the telco
needs to know that you want FGD.
AND...
If you are connecting to an Access Tandem instead of and End Office, then
the featdmf in Asterisk will not work.
I have submitted a request for a quote to Digium to modify the code to make
this work properly.
Likewise, true FGB terminating (where it looks like you are the carrier)
works through an Access Tandem and the additional code is missing for that
also.
Take out the featb and add:
em_w
This will let you see if just plain old DTMF works.
James Taylor
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Ackley
Sent: Sunday, April 24, 2005 4:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] T1 EM false busy after dial
TE101P card T1 EM trunk to telco
on a SIP-PSTN call, after dial
SIP phone hears two seconds busy tone (1) then ring tone
how do we get rid of busy tone?
(1) two second busy
(480+620/500 0/500 480+620/500 0/500)
---
extensions.conf:
;
; dial-out to the PSTN with 7 digits
;
exten = _NXX,1,Dial(Zap/g1/${EXTEN})
exten = _NXX,n,Hangup()
zaptel.conf:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
zapata.conf:
[trunkgroups]
[channels]
language=en
context=default
signalling=featb
usecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=8
channel = 1-24
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[Asterisk-Users] Zaphfc problem
I upgraded my FC3 to kernel 2.6.11. I installed bristuff 0.2.0-RC8 and I cannot call out using zaphfc. I can receive calls, but can't get out. Here is what I got: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, Zap/1/**348|60|rTt) in new stack [ 00 e7 0e 12 08 01 04 05 04 03 80 90 a3 18 01 81 6c 05 41 80 31 30 31 70 0b c1 30 35 30 31 33 35 39 33 34 38 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115EA: 1 N(S): 007 0: 0 N(R): 009 P: 0 33 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 1 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 81] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 ChanSel: B1 channel ] [6c 05 41 80 31 30 31] Calling Number (len= 7) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '101' ] [70 0b c1 *** 33 34 38] Called Number (len=13) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '**348' ] [a1] Sending Complete (len= 1) -- Called 1/**348 [ 00 e7 01 10 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 008 P/F: 0 0 bytes of data -- ACKing all packets from 6 to (but not including) 8 -- ACKing packet 7, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter [ 02 e7 12 10 08 01 84 02 18 01 89 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115EA: 1 N(S): 009 0: 0 N(R): 008 P: 0 7 bytes of data -- ACKing all packets from 7 to (but not including) 8 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: CALL PROCEEDING (2) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Sending Receiver Ready (10) [ 02 e7 01 14 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 010 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ 02 e7 14 10 08 01 84 45 08 02 82 95 1c 0f 91 a1 0c 02 02 16 c8 06 06 04 00 87 69 01 07 1e 02 82 88 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115EA: 1 N(S): 010 0: 0 N(R): 008 P: 0 29 bytes of data -- ACKing all packets from 7 to (but not including) 8 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=29 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: DISCONNECT (69) [08 02 82 95] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) ] [1c 0f 91 a1 0c 02 02 16 c8 06 06 04 00 87 69 01 07] Facility (len=17, codeset=0) [ 0x91, 0xa1, 0x0c, 0x02, 0x02, 0x16, 0xc8, 0x06, 0x06, 0x04, 0x00, 0x87, 'i', 0x01, 0x07 ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Sending Receiver Ready (11) [ 02 e7 01 16 ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 011 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Channel 0/1, span 1 got hangup -- Zap/1-1 is circuit-busy [ 00 e7 10 16 08 01 04 4d 08 02 81 95 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115EA: 1 N(S): 008 0: 0 N(R): 011 P: 0 8 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 4/0x4) (Originator) Message type: RELEASE (77) [08 02 81 95] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Call Rejected (21), class = Normal Event (1) ]
Re: [Asterisk-Users] signate.com webcall
On 20 Apr 2005, at 17:12, Moody wrote: Signate offers an interesting product they call 'webcall', which basically contacts a client at a number they provide then connects that person to a sales staff. Some potential for abuse but a nice idea for support etc. I know that it is possible to do (obviously) and well documented but has anyone actually released an open product similar to signate's webcall or even a basic web initiated call interface (ie for calling cards). I wasn't able to track via google or the wiki any ongoing projects - is anyone interested in working on something like this? J You can get this, is a little remake maybe you can use. http://www.asteriskspain.org publish in download section free php webcall Adrià Vidal xpreme.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy and Debian
On Sun, Apr 24, 2005 at 01:15:06PM +0200, Thore wrote: Hi ! What is the easiest esyest way for implementation of ztdummy on a Debian (testing) system? Thore On testing/unstable you basically: apt-get install module-assistant m-a a-i zaptel to build a zaptel package for your kernel. You need its kernel-headers package installed. If you're sane and avoid building as root: As root: apt-get install zaptel-source module-asistant As user: m-a build -u . -t build zaptel A zaptel-modules-deb package will be generated in the same directory which you can install as root. And some binary packages for the sarges: http://tzafrir.org.il/rapid/ -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztcfg doesn't do anything from /etc/rc.d/rc.local
On Sun, Apr 24, 2005 at 06:45:15AM +0200, Remco Barende wrote: When using bristuff I do get an error too if I don't load zaptel first but not with the tdm driver. I know that in my modprobe.conf it is specified that ztcfg should be run after loading the module but why doesn't it? That's a good question. cat EOF /usr/local/sbin/ztcsfg_trace #!/bin/sh exec -o /tmp/ztcfg.trace /sbin/ztcfg $@ EOF chmod 755 /usr/local/sbin/ztcfg_trace Now edit that modprobe file and use /usr/local/sbin/ztcfg_trace instead of ztcfg. Does it get executed? (does the file /tmp/ztcfg.trace get generated?) What happens? For some reason ztcfg is only 'accepted' when run from the cli One wild guess: someone uses the name 'ztcfg' and /sbin/ztcfg is not on the PATH when run from the init script? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfers fails, even after upgrade to 1.0.7
Hi We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected to old PBX, and some SIP phones, used by a callcenter with queues. Almost all calls are incoming (through E1 line), answered by some callcenter operator (using SIP phones, call assigned by queue app), and in some cases, are transferred to some other extension on the old PBX or other SIP. We had problems with Music on Hold (on the queue) and with transfers on version 1.0.3. We now upgraded to 1.0.7 and the MoH problem is gone, but we still have some transfer problems. What happens is that sometimes when one callcenter op (SIP client) does a transfer to another SIP or an extension that is mapped to a FXO line (old PBX), we get a half-call: the caller hears the called station, but the called station (the one the call is transferred to) does not here the caller. As we need attended transfer, the calls are made from the SIP phone (Xten), using the transfer button (not blind transfers). Don't really know how to debug this. Is there a log I can see that can help me pinpoint the problem?. On that log, what should we be looking for? I'm used to debug this kind of problems in general, but are not familiar with SIP protocol nor Asterisk debugging. We tried to change SIP phones, but its the same. Note that it happens with calls that have one end on the E1 and the other to FXO, both local to Asterisk (joined by a SIP phone), so it does not seems to be a codec problem. Thanks for any advice. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] signate.com webcall
[EMAIL PROTECTED] wrote: On 20 Apr 2005, at 17:12, Moody wrote: Signate offers an interesting product they call 'webcall', which basically contacts a client at a number they provide then connects that person to a sales staff. Some potential for abuse but a nice idea for support etc. I know that it is possible to do (obviously) and well documented but has anyone actually released an open product similar to signate's webcall or even a basic web initiated call interface (ie for calling cards). I wasn't able to track via google or the wiki any ongoing projects - is anyone interested in working on something like this? J You can get this, is a little remake maybe you can use. http://www.asteriskspain.org publish in download section free php webcall What is the Spanish word for download Is the program also in Spanish??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI problem on Zaptel channel
I found if put AGI on zaptel channel, when execute stream file there is no voice and execute set callerid got no effect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] signate.com webcall
Bookmark this page .. It has saved me more than once in dealing with pages with different languages .. http://babelfish.altavista.com/babelfish/ BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Sunday, April 24, 2005 5:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] signate.com webcall [EMAIL PROTECTED] wrote: On 20 Apr 2005, at 17:12, Moody wrote: Signate offers an interesting product they call 'webcall', which basically contacts a client at a number they provide then connects that person to a sales staff. Some potential for abuse but a nice idea for support etc. I know that it is possible to do (obviously) and well documented but has anyone actually released an open product similar to signate's webcall or even a basic web initiated call interface (ie for calling cards). I wasn't able to track via google or the wiki any ongoing projects - is anyone interested in working on something like this? J You can get this, is a little remake maybe you can use. http://www.asteriskspain.org publish in download section free php webcall What is the Spanish word for download Is the program also in Spanish??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Static and echo on PRI
I need some serious help!! I have been in the process of building an
Asterisk system to replace a Cisco Call Manager. I have most everything
setup, but only got to test the PRI today. To make a long story short,
my Call Manager is half broken and I need to go live with * a lot sooner
than I expected.
Here's where I am and what I tried. I am using all Cisco phones, mostly
7940's and 60's in a SIP configuration. All internal calls work with no
issues. I have a TE405P for the PRI and a TDM22B for my paging system
and whatnot. I am currently only using one PRI on the quad card. When
calling out on the PRI, I am getting static and some echoing. I have
tried various orders and values for the txand rxgains, echocancellation
and nothing seems to help. I get the staticy noise only when sound is
coming in, like when the other is ringing or when the other person is
talking. Complete silence the rest of the time. I get different
amounts of echo when calling out, the person on the other end says they
hear no echo or static at all, just on the SIP phones. I made sure that
I have no IRQ conflicts (output below) and my CPU usage seems to be
fine, plenty of horsepower remaining.
Here are the parts of my configs that I feel are relavent:
/etc/zaptel.conf
---
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
fxoks=97
fxoks=98
fxsks=99
fxsks=100
/etc/asterisk/zapata.conf
--
[trunkgroups]
[channels]
context=incoming
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
immediate=no
callerid=xx
rxgain=0.0
txgain=0.0
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;echotraining=800
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-23
signalling=fxo_ks
group = 2
channel = 97
signalling=fxo_ks
group = 3
channel = 98
signalling=fxs_ks
group = 4
channel = 99
signalling=fxs_ks
group = 5
channel = 100
/etc/asterisk/extensions.conf
---
TRUNK=Zap/g1
TRUNKMSD=1
[trunklocal]
exten = _6NX,1,SetCallerID(xx)
exten = _6NX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _6NX,3,Congestion
Here are a few lines from the logs that might mean something to someone:
Apr 24 18:22:40 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:22:40 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:22:41 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:22:41 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:24:31 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:25:53 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:25:54 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:09 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:10 WARNING[196620]: PRI: !! Got reject for frame 51, but we
only have others!
Apr 24 18:26:10 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:26:11 NOTICE[196620]: PRI got event: 8 on Primary D-channel of
span 1
Apr 24 18:27:01 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
Apr 24 18:27:41 NOTICE[196620]: PRI got event: 6 on Primary D-channel of
span 1
** I tried the line span=1,0,0,esf,b8zs in my zaptel.conf and made no
difference.
Here is a debug section for my PRI when I was getting static and echo:
Enabled debugging on span 1
-- Executing SetCallerID(SIP/226-9fca, 3307551414) in new stack
-- Executing Dial(SIP/226-9fca, Zap/g1/3305596313) in new stack
-- Making new call for cr 32771
Protocol Discriminator: Q.931 (8) len=46
Call Ref: len= 2 (reference 3/0x3) (Originator)
Message type: SETUP (5)
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
Ext: 1 User information layer 1: u-Law (34)
[18 03 a9 83 81]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: User (0)
Ext: 1 Progress Description: Calling
equipment is non-ISDN. (3) ]
[6c 0c 21 80 33 33 30 37 35 35 31 34 31 34]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Re: [Asterisk-Users] signate.com webcall
Hello- I made some adjustments to the Ast-Tapi to do a similar thing on my site. It was a very easy modification. Here is a sample running on our demo server. I would appreciate it if people don't just try it though -- since the calls are routed to my sales staff who I pay per call... heh. http://crm.yarnia.com:81/cgi-bin/taci.pl I will be happy to send my changes to anyone that asks, email me off list. To joseph @ yarnia dot net -- NOT to this address (my listserv dump). Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Mark Johnson wrote: I need some serious help!! I have been in the process of building an Here are my interrupts: cat /proc/interrupts CPU0 0: 960018 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3:9565339 XT-PIC t4xxp 5: 16301 XT-PIC eth0 8: 1 XT-PIC rtc 9: 0 XT-PIC usb-ohci, usb-ohci, usb-ohci, ehci_hcd 10: 0 XT-PIC ohci1394 11:9566436 XT-PIC wctdm 12: 23 XT-PIC PS/2 Mouse 14: 16978 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 Try 'vmstat 1'--are you getting 40% system utilization every n seconds? If so, unload the wcfxo and wcfxs modules and test again. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 EM false busy after dial
Normally, plain old PBX DID trunks are em_w (dtmf).
Strange, the only other problem might be the timing of the wink.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Ackley
Sent: Sunday, April 24, 2005 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 EM false busy after dial
thanks info and suggestion
we have a plain old PBX DID trunk from our telco
will try to get more information about the trunk
meanwhile I tried as documented in my zapata.conf:
; JNA tried all below - and even NO signaling same resuts
;Apr 24 21:11:15 WARNING[4430]: chan_zap.c:10198 setup_zap: Ignoring
:signalling
;-- Reconfigured channel 1, Feature Group B (MF) signalling
; etc.
;
;signalling=featb
:signalling=em_w
;signalling=sf_featb
;signalling=sf_featdmf
;signalling=sf
jltaylor wrote:
If Feature Group B signaling is working properly (and you have Feature
Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is
1
or 0 based on the number assigned to you}.
If you are dialing out {terminating where you look like the carrier} on
FGB then it depends on if you are connected to an Equal Access End Office
or
a Access Tandem.
Are you sure about the Feature Group B thing or do you have trunks that
just
require MF signaling?
If you want MF, you might try the featdmf setting, however, the telco
needs to know that you want FGD.
AND...
If you are connecting to an Access Tandem instead of and End Office, then
the featdmf in Asterisk will not work.
I have submitted a request for a quote to Digium to modify the code to make
this work properly.
Likewise, true FGB terminating (where it looks like you are the carrier)
works through an Access Tandem and the additional code is missing for that
also.
Take out the featb and add:
em_w
This will let you see if just plain old DTMF works.
James Taylor
903-793-1956
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[Asterisk-Users] Asterisk best practices
List, I have been using asterisk for a couple of weeks now, to support some Cisco 7960 and 7920 phones, and have been enjoying the learning experience. I have gotten the phone firmware upgraded, Broadvoice connectivity, basic dial plan, and voice mail working. However I am sure that there is more that I can do. So my question, what is the best feature of Asterisk, and how have you deployed it in your organization? What trick configuration have you come up with to do something really out of the box cool? If you can document it with come configuration samples, so much the better. Thanks in advance Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Michael Welter wrote: Try 'vmstat 1'--are you getting 40% system utilization every n seconds? If so, unload the wcfxo and wcfxs modules and test again. I tested and I do in fact get from 40-50% system util every 5 seconds or so. After removing the wctdm module, the system util drops to 0 and stays there. I have not loaded the wcfxs and wcfxo modules because I could never get them to work right. I instead load the wctdm and it has seemed to work fine. I only need to make the fx port to the paging system work and the others can stay idle. What modules and order so you suggest. Here is what I load in this order: wct4xxp wctdm Thanks! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
i am trying to get G723 passthrough get the same error. how to configure passthrough for g723/g729 ? On 4/24/05, Brian Capouch [EMAIL PROTECTED] wrote: jltaylor wrote: ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. Still no joy. Added the allow=g729 to general, too, and I still get the same errors. Thanks anyways. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with gotoiftime and cvs head
Anybody having problems with cvs head?
I gave a problem with queues and agents... I have defined joinempty=no on
queues.conf and eventhough there are no agents logged in, the call are
getting queued.
Also, I have the following statements:
exten = s,9,GotoIfTime(00:00-11:59|*|*|*?10:12)
exten = s,10,Background(${SONIDOS}/buenosdias)
exten = s,11,Goto(18)
exten = s,12,GotoIfTime(12:00-18:59|*|*|*?13:15)
exten = s,13,Background(${SONIDOS}/buenastardes)
exten = s,14,Goto(18)
exten = s,15,GotoIfTime(19:00-23:59|*|*|*?16:18)
exten = s,16,Background(${SONIDOS}/buenasnoches)
exten = s,17,Goto(18)
exten = s,18,GotoIfTime(9:00-19:00|mon-fri|*|*?20:19)
exten = s,19,Background(${SONIDOS}/horariooficinas)
exten = s,20,Background(${SONIDOS}/intruder-bienvenida)
Now its 9.30 pm and when a call comes in, the call is played the good
morning message, any ideas why?? This beats me
[EMAIL PROTECTED] asterisk]# date
Sun Apr 24 21:48:13 CDT 2005
-- Executing SetCIDName(IAX2/[EMAIL PROTECTED], Intruder:
202) in new stack
-- Executing GotoIfTime(IAX2/[EMAIL PROTECTED],
00:00-11:59|*|*|*?10:12) in new stack
-- Executing BackGround(IAX2/[EMAIL PROTECTED],
/var/lib/asterisk/sounds/akrall/buenosdias) in new stack
-- Playing '/var/lib/asterisk/sounds/akrall/buenosdias' (language
'default')
-- Playing '/var/lib/asterisk/sounds/akrall/buenosdias' (language
'default')
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RE: [Asterisk-Users] Asterisk best practices
Ditto! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Simon Sent: Sunday, April 24, 2005 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk best practices List, I have been using asterisk for a couple of weeks now, to support some Cisco 7960 and 7920 phones, and have been enjoying the learning experience. I have gotten the phone firmware upgraded, Broadvoice connectivity, basic dial plan, and voice mail working. However I am sure that there is more that I can do. So my question, what is the best feature of Asterisk, and how have you deployed it in your organization? What trick configuration have you come up with to do something really out of the box cool? If you can document it with come configuration samples, so much the better. Thanks in advance Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What software and types of connections are used by VOIP providers
I would like to start a discussion about real and big voip providers. Lets say for example voicepulse or vonage or any other. What software do they use? Is anybody using asterisk? What kind of connections they have to the internet? What kind of equpment is used by them? I hope that this group can be used for such a topic. If not, please don't replay, or direct me to other groups. Thanks Bart, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with gotoiftime and cvs head
I figured out the problem with gotoiftime.. Still have the problem with the
queues though :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 24 de Abril de 2005 09:45 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problems with gotoiftime and cvs head
Anybody having problems with cvs head?
I gave a problem with queues and agents... I have defined joinempty=no on
queues.conf and eventhough there are no agents logged in, the call are
getting queued.
Also, I have the following statements:
exten = s,9,GotoIfTime(00:00-11:59|*|*|*?10:12)
exten = s,10,Background(${SONIDOS}/buenosdias)
exten = s,11,Goto(18)
exten = s,12,GotoIfTime(12:00-18:59|*|*|*?13:15)
exten = s,13,Background(${SONIDOS}/buenastardes)
exten = s,14,Goto(18)
exten = s,15,GotoIfTime(19:00-23:59|*|*|*?16:18)
exten = s,16,Background(${SONIDOS}/buenasnoches)
exten = s,17,Goto(18)
exten = s,18,GotoIfTime(9:00-19:00|mon-fri|*|*?20:19)
exten = s,19,Background(${SONIDOS}/horariooficinas)
exten = s,20,Background(${SONIDOS}/intruder-bienvenida)
Now its 9.30 pm and when a call comes in, the call is played the good
morning message, any ideas why?? This beats me
[EMAIL PROTECTED] asterisk]# date
Sun Apr 24 21:48:13 CDT 2005
-- Executing SetCIDName(IAX2/[EMAIL PROTECTED], Intruder:
202) in new stack
-- Executing GotoIfTime(IAX2/[EMAIL PROTECTED],
00:00-11:59|*|*|*?10:12) in new stack
-- Executing BackGround(IAX2/[EMAIL PROTECTED],
/var/lib/asterisk/sounds/akrall/buenosdias) in new stack
-- Playing '/var/lib/asterisk/sounds/akrall/buenosdias' (language
'default')
-- Playing '/var/lib/asterisk/sounds/akrall/buenosdias' (language
'default')
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Re: [Asterisk-Users] Static and echo on PRI
Mark Johnson wrote: Michael Welter wrote: Try 'vmstat 1'--are you getting 40% system utilization every n seconds? If so, unload the wcfxo and wcfxs modules and test again. I tested and I do in fact get from 40-50% system util every 5 seconds or so. After removing the wctdm module, the system util drops to 0 and stays there. I have not loaded the wcfxs and wcfxo modules because I could never get them to work right. I instead load the wctdm and it has seemed to work fine. I only need to make the fx port to the paging system work and the others can stay idle. What modules and order so you suggest. Here is what I load in this order: wct4xxp wctdm Do you still have the static on the PRI without the TDM modules? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why can't I hear audio?
Hi Everybody can someone tell me why I can hear audio? My call is to my proxie which is directing it to my Asterisk box. The Voice mail is playing but I think its playing to my proxie. the phone is on 198.31.185.246:63257 Here is from the sip debug. Thanks Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:63257;user=phone Supported: replaces Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, qop=auth, nc=0001, cnonce=1a605453cf8a557d, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=874d55e7960ad550b78bb1d8660faf69 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 338 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 asterisk1*CLI v=0 o=6262769011 8000 8001 IN IP4 198.31.185.246 s=SIP Call c=IN IP4 198.31.185.246 t=0 0 m=audio 63268 RTP/AVP 0 4 9 15 2 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:15 G728/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 16 headers, 15 lines Using latest request as basis request Sending to 208.41.254.119 : 5060 (non-NAT) Found no matching peer or user for '208.41.254.119:5060' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 198.31.185.246:63268 Found description format PCMU Found description format G723 Found description format G722 Found description format G728 Found description format G726-32 Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115 (g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 9009 in from-sip-external list_route: hop: sip:208.41.254.119;lr;hash=sipd-0-2-2 list_route: hop: sip:[EMAIL PROTECTED]:63257;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.41.254.119:5060 -- Executing VoiceMail(SIP/208.41.254.119-089aef50, 9009) in new stack We're at 208.41.254.125 port 13630 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 v=0 o=root 2330 2330 IN IP4 208.41.254.125 s=session c=IN IP4 208.41.254.125 t=0 0 m=audio 13630 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 208.41.254.119:5060 -- Playing 'vm-intro' (language 'en') asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQAOCCEtoOY6oOebox7ZBwoRRiY_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK2e698f7c1c332d46 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Contact: sip:[EMAIL PROTECTED]:63257;user=phone Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0002, cnonce=b85d4240018f156a, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=4030f97656e76c9bffecee6942efbfcc Call-ID: [EMAIL PROTECTED] CSeq: 55676 ACK User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow:
Re: [Asterisk-Users] Asterisk best practices
Some things i've wanted to look into is sphinx2 with asterisk. They have EAGI demos but surely they don't work for me! How about voicemail linked to your online activity (aim) http://ruk.ca/article/1832 . Dream it and write it. On 4/24/05, Brian Watters [EMAIL PROTECTED] wrote: Ditto! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Simon Sent: Sunday, April 24, 2005 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk best practices List, I have been using asterisk for a couple of weeks now, to support some Cisco 7960 and 7920 phones, and have been enjoying the learning experience. I have gotten the phone firmware upgraded, Broadvoice connectivity, basic dial plan, and voice mail working. However I am sure that there is more that I can do. So my question, what is the best feature of Asterisk, and how have you deployed it in your organization? What trick configuration have you come up with to do something really out of the box cool? If you can document it with come configuration samples, so much the better. Thanks in advance Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
Asking someone to spend eleven times more money on a hardware purchase ($220 vs $20) as a gesture of good will is expecting a bit much, I'd say. Certainly I can understand that Digium doesn't stand to make much money selling X100Ps at $10 each, and I can certainly understand them choosing to not sell them. But, by the same token I cannot understand the community's interest in discouraging other folks from joining the community in the way that economically suits them best. Lee. On Sun, 24 Apr 2005, Dean Collins wrote: Don't forget you may like to support digium by buying an official tdm400P I know more expensive then a $7 clone but will work better on lines different to the 600ohm US pstn Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 4:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Excellent news! Now, remember that I am in Europe and out of my Linksys VoIP router I have a phone line coming out which I believe is North American standard. I am really not knowledgeable enough about the differences in both Networks so do I need a special card for Europe ? Otherwise, I'm set to purchase 2 X100P cards! The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can Asterisk do the following for me ?
Have you check on what price a nec ip pabx is going for lately? Whilst I appreciate that digium should be selling their cards for less - if there was no digium there would be no asterisk - therefore price of clones x00p's is irrelevant. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Sunday, April 24, 2005 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Asking someone to spend eleven times more money on a hardware purchase ($220 vs $20) as a gesture of good will is expecting a bit much, I'd say. Certainly I can understand that Digium doesn't stand to make much money selling X100Ps at $10 each, and I can certainly understand them choosing to not sell them. But, by the same token I cannot understand the community's interest in discouraging other folks from joining the community in the way that economically suits them best. Lee. On Sun, 24 Apr 2005, Dean Collins wrote: Don't forget you may like to support digium by buying an official tdm400P I know more expensive then a $7 clone but will work better on lines different to the 600ohm US pstn Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 4:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Excellent news! Now, remember that I am in Europe and out of my Linksys VoIP router I have a phone line coming out which I believe is North American standard. I am really not knowledgeable enough about the differences in both Networks so do I need a special card for Europe ? Otherwise, I'm set to purchase 2 X100P cards! The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] Can Asterisk do the following for me ?
What year is this? 2005 right? Doesn't everyone on the planet know that you get what you pay for these days? If you want to experiment with Asterisk there is nothing wrong with using clone X100P cards at $6.95 a pop. If you are putting in a production machine that is mission critical to the operation of a company, do you want your entire phone system to be dependant on a $7 card? You would want a high quality card that comes complete with technical support. THAT'S when it makes complete sense to fork over some cash for a quality piece of equipment. If you are really diving into Asterisk, you would probably want to get the developer's kit just so you are working with equipment that you will most likely be using in a production environment. For us, our demo systems and backup systems run clone cards but our production systems all use Digium cards. Kerry Garrison http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Sunday, April 24, 2005 9:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Have you check on what price a nec ip pabx is going for lately? Whilst I appreciate that digium should be selling their cards for less - if there was no digium there would be no asterisk - therefore price of clones x00p's is irrelevant. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Sunday, April 24, 2005 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Asking someone to spend eleven times more money on a hardware purchase ($220 vs $20) as a gesture of good will is expecting a bit much, I'd say. Certainly I can understand that Digium doesn't stand to make much money selling X100Ps at $10 each, and I can certainly understand them choosing to not sell them. But, by the same token I cannot understand the community's interest in discouraging other folks from joining the community in the way that economically suits them best. Lee. On Sun, 24 Apr 2005, Dean Collins wrote: Don't forget you may like to support digium by buying an official tdm400P I know more expensive then a $7 clone but will work better on lines different to the 600ohm US pstn Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 4:08 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Can Asterisk do the following for me ? Excellent news! Now, remember that I am in Europe and out of my Linksys VoIP router I have a phone line coming out which I believe is North American standard. I am really not knowledgeable enough about the differences in both Networks so do I need a special card for Europe ? Otherwise, I'm set to purchase 2 X100P cards! The short answer is Yes. However, you would need X100P cards and not regular modem cards. These cards can be found on eBay for about $7 each. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, April 24, 2005 12:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can Asterisk do the following for me ? Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also,
[Asterisk-Users] Failed to authenticate
HI,all!
I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
context=defaulttos=0x18dbname=asteriskdbhost=localhostdbuser=asteriskdbpass=password
extensions.conf
[general]static=yeswriteprotect=no
[globals]CONSOLE=Console/dsp
[local]
exten = _X.,1,Dial(SIP/${EXTEN},20,t)exten = _X.,2,Hangup
[default]include = demoinclude = local
I have also setted callidnum 1000-1010 in mysql database.First,it can dial out and receive a call well.(in internal) then I alter callidnum 1000 to
1000.It can registered successfully and it can receive a call ,but it cannot dial out .There are some words in my asterisk console:"Failed to authenticate user "aaa" sip:[EMAIL PROTECTED]; tag=164262242".So,I tried change callidnum to 1000, it works. I don't know what happen.Can anybody tell me what's the matter ? thanks!
in addition,If I don't use sipfriends with mysql, it does well !
Do You Yahoo!?
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[Asterisk-Users] Trouble with call parking/transfer
Hi all, I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. Before I go ripping things apart, I was really wondering if this is by design, or should it work on all my devices? I have an iaxy, phones hanging off fxs ports on a pair of tdm400p's, a sipura 841, a sipura 3000, and a pair of sipura 2000's and a Polycom IP 500. It only works on the phones hanging off the tdm400p. Should this work on all phones? Does anyone have it working on non digium FXS phones? Thanks, Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
Hello, that is even possible without MODEM hardware. It should work with a simple call forwarder/diverter. It connects to both line ends and works more or less like a analogue 2-port pbx with a fixed programmable forwarding number. Offered e.g. in Germany from AUERSWALD (A-BOX) http://www.auerswald.de/int/products/auerswald_box/box_intro.htm or at EBAY ...like here http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemitem=6386901484 No modems or VoIP equipment except the ATA is needed at all for this ... regards, Jürgen Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
Franz wrote: Please contact me Urgent... Hi Frantz, I can do custom programming. Here is some information about my company: http://ykoz.net/intl/ Let me know what you're after and I'll send you a preliminary quote. Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
