RE: [asterisk-users] Does Asterisk support DNIS?
From: David Ruggles [EMAIL PROTECTED]
Date: Sun, 18 Feb 2007 20:41:46 -0500
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a
difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
extension number so most of the dialplain suggestions offered so far won't
work. I did try the Wait() function as was suggested. I tried it first in
an
s extension but this didn't work, it still gave the error: Unknown
extension '1' in context '1st-T1' requested I then changed it to extension
1 and while it does seem to work (it doesn't try the other extensions) it
seems like the DNIS is completely lost.
I'm still confused. As others indicated, when DNIS is offered, there must
be a delimiter of some kind (like *). When both ANI and DNIS are offered,
the order must be predetermined, too.
I understand that your dial plan reacts to each digit as separate
extensions. The first thought I have is: what's the TIMEOUT(digits) value?
Might it be too short so Asterisk won't see the DNIS string as a group?
A second thought test is: do you see any of the predefined delimiters in
Asterisk? i.e., when dial plan branches to DNIS digits, does it ever stop at
extension '*'?
DNIS must use either timeout and delimiter, or both, to separate itself from
actual voice.
Assume delimiters are there, and assume that your dial plan won't budge with
a reasonable TIMEOUT(digits) so it still sends the call to extensions '*',
'1' - '5', '*'. There are still ways to reassemble the entire DNIS string
from these discrete extensions. e.g., (untested code)
exten = s,1,DNIS=; resets
exten = s,n,Answer()
exten = *,1,GotoIf(${DNIS}=?more:); delimited as *DNIS*
exten = *,n,DNIS=${DNIS:1}
exten = *,n,Goto(handleDNIS,${DNIS},1); hand call to another context
exten = *,n(more),NoOp(start collecting DNIS)
exten = _X,1(),DNIS=${DNIS}${EXTEN}
Yuan Liu
As I said in my first post (although it may have been a little too
abrasive)
this configuration is very standard and so I find it hard to believe that
Asterisk can't handle it.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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RE: [asterisk-users] Summary of Trixbox vs. custom install
You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? /snip Please note that the recent (2.x) releases of trixbox allow you to select which modules to install, including raid. Really? I didn't find any information about ths feature... Could you tell me how to install on RAID with the 2.0 installer? Which option should I pass to the installer? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Summary of Trixbox vs. custom install
At 20.13 18/02/2007, you wrote: You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Very useful information. Thanks! Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? I agree. There's probably a choice behind it, but it's obscure to me. 2) How easy it is to find Trixbox SRPMS? Is it possible to compile new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without having to rewrite all the configuration files, changing all paths, all permissions, and so on... You can update Asterisk/zaptel/whatever by just downloading the source and compiling it. My home system was installed with [EMAIL PROTECTED] on version 0.8 (if memory serves me right) and I upgraded Asterisk and Zaptel to version 1.0.10 by downloading and compiling. I know, this is a really old version and I should upgrade, but hey, it is doing everything I need and it is stable (uptime of 315 days). You mean compiling raw tar.gz or SRPMS? And where do you download them from? Trixbox site or the original vendors' sites? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problem with busydetect and cell phones
Ryan McDaniel wrote: I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal correctly and clears the line. If I call from an internal extension to a cell phone and then hangup the cell phone Asterisk will never detect the busy signal though it is clearly there. Asterisk will happily sit there listening to the busy signal. I suspect that the busy signal styles are slightly different though it is undetectable to me. How can I fix this??? It causes severe issues when a call is forwarded to a cell phone via the Zap interfaces as once you hangup the cell phone Asterisk never releases the channel. The landlines are with ATT. The cell phones I'm testing with are Cingular (ATT subsidiary). There must be a subtle difference in the busy signals. How can I make it catch busy signals from both carriers? Have you tried calling ATT and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific circuit in software. Particularly if you have a business line -- you can demand it. All PBXs need it if they use analog lines (and plenty still do) so I'm sure this is not an alien concept to ATT. It's just a matter of finding the right Earthling there who can help you. This might be one of those times where a beer with the technician will get you some joy, if calling Repair doesn't give you any joy. -Stephen- Unfortunately I tried that. Apparently my lines are on one of the last really ancient junction boxes in Southern California. When using busydetect is it looking for any on / off repetitive sound to identify the busy signal, or for a specific length sound as defined in the indications.conf region? I'd really like to avoid using callprogress if possible. Is there a way to tweak it so it will accept a wider variety of busy patterns? - Ryan Ryan, Even 1AESS switches offer disconnect supervision -- and I am not aware of any of those still in primary service in Southern California. By early 2000, Pacific Bell (then SBC, now ATT) replaced all the analogue 1As with DMS-100s. If you care to contact me off list, I may be able to help get you in touch with the right department to assist you. Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinct call permissions for each user
Thanks Luki, that's exactly what I was looking for, I'll give it a try...
Regards,
Ricardo.
Luki wrote:
someone please give me one example?
[locals]
exten = _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten = _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
exten = s,2,Dial(Zap/.../${ARG1})
[fullaccess]
include = locals
include = longdistance
include = ...
[restricted]
include = locals
include = ...
Put user A into the restricted context, and user B into the fullaccess
context. You can include other extension (i.e. services) and implement
roll-over onto a backup trunks in macro-outcall.
You can of course also simply it and only have two contexts and no
macro, etc.
--Luki
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[asterisk-users] Problems with CentOS ztdummy kernel 2.6
Hi List, I am having some trouble with installing the latest version of ztdummy on a CentOS Kernel 2.6 system. I have installed a few Asterisk systems on Slackware Kernel 2.4.x without any issues, unfortunately there is no choice about this distro, or kernel as it has been preinstalled by someone else. And so I am in the dark with an unfamiliar distro and kernel. I am fairly sure the kernel source has been installed. I'm not sure the timer module is installed in the kernel, is it possible to check? If not I think I will need to use ztdummy for definite. Any help with this would be a real life saver. Thanks - Chris From the zaptel-1.2.13 directory I issue the make linux26 command with the following result: make: *** No rule to make target `linux26'. Stop. Just issuing the make command does seem to work and concludes with: make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Make install outputs the following: make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 HOTPLUG_FIRMWARE=yes modules make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ install -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ install -m 644 wctc4xxp/*.bin /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ install -m 644 wct4xxp/*.ima /lib/firmware; \ install -m 644 wctc4xxp/*.bin /lib/firmware; \ fi Installed firmware install -D -m 755 ztcfg /sbin/ztcfg if [ -f sethdlc-new ]; then \ install -D -m 755 sethdlc-new /sbin/sethdlc; \ elif [ -f sethdlc ]; then \ install -D -m 755 sethdlc /sbin/sethdlc ; \ fi if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi for x in zaptel.ko tor2.ko torisa.ko wcusb.ko wcfxo.ko wctdm.ko wctdm24xxp.ko ztdynamic.ko ztd-eth.ko wct1xxp.ko wcte11xp.ko pciradio.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/extra/$x ; \ done; \ make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' INSTALL /usr/src/zaptel-1.2.13/pciradio.ko INSTALL /usr/src/zaptel-1.2.13/tor2.ko INSTALL /usr/src/zaptel-1.2.13/torisa.ko INSTALL /usr/src/zaptel-1.2.13/wcfxo.ko INSTALL /usr/src/zaptel-1.2.13/wct1xxp.ko INSTALL /usr/src/zaptel-1.2.13/wct4xxp/wct4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctc4xxp/wctc4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctdm.ko INSTALL /usr/src/zaptel-1.2.13/wctdm24xxp.ko INSTALL /usr/src/zaptel-1.2.13/wcte11xp.ko INSTALL /usr/src/zaptel-1.2.13/wcusb.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxo.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxs.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp_usb.ko INSTALL /usr/src/zaptel-1.2.13/zaptel.ko INSTALL /usr/src/zaptel-1.2.13/ztd-eth.ko INSTALL /usr/src/zaptel-1.2.13/ztd-loc.ko INSTALL /usr/src/zaptel-1.2.13/ztdummy.ko INSTALL /usr/src/zaptel-1.2.13/ztdynamic.ko INSTALL /usr/src/zaptel-1.2.13/zttranscode.ko make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' if ! [ -f wcfxsusb.o ]; then \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxsusb.o; \ fi; \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxs.o install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 [ `id -u` = 0 ] /sbin/ldconfig || : rm -f /usr/lib/libtonezone.so ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then /sbin/restorecon -v /usr/lib/libtonezone.so; fi install -D -m 644 zaptel.h /usr/include/linux/zaptel.h install -D -m 644 torisa.h /usr/include/linux/torisa.h install -D -m 644 tonezone.h /usr/include/tonezone.h install -m 644 doc/ztcfg.8 /usr/share/man/man8 install -m 644 doc/zttool.8 /usr/share/man/man8 [ `id -u` = 0 ] /sbin/depmod -a 2.6.9-42.0.3.ELsmp || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc wct4xxp wcfxs wctdm8xxp wct2xxp Building /etc/modprobe.conf... Once it is installed I run: modprobe ztdummy with the following result. FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Asterisk and a modem pool.
Asterisk and a modem pool. I have a small modem pool, after its carry for an asterisk modems have ceased will incorporate or incorporate, but for the speed 9600. I use Tormenta 2b2. Prompt the decision. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PPPD with analog lines
Hi All, Is it possible to use asterisk as a internet link backup callback solution? I mean when my main DSL link is down at my server room I would like to dial to asterisk , then it will call back me and provide a connection to a LAN network. Regards, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support DNIS?
You can have multiple control channels on a PRI.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
On Sun, 18 Feb 2007, Matt wrote:
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have
all
the intelligence you need to do it right.
You may not have that option. For example, you want to split a T1 from a
legacy PBX to 12 channels to a proprietary IVR system and 12 channels to
an Asterisk box. Can't do that with with PRI and a single T1 because you
only have one control channel.
--Ron
On 2/18/07, Matt [EMAIL PROTECTED] wrote:
Why would the card care? This would be something you'd take care of
in
your dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been
mentioned
but DNIS and ANI can be variable length without confusion if the
sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS*
(pronounced
Star ANI Star DNIS Star allows the receiver to identify the two
values
unambiguously and to find the trailing boundary (when the 3rd *
has
been
received).
We have a Channelized Voice T1 from a long distance provider that is
set
up this way into our non-Asterisk PBX where the provider sends us
ANI as
the full originating phone number and DNIS as the last 4
digits. So
the
DTMF string seen by our PBX for someone calling one of our toll-free
numbers, say 800-123-4567, from a local phone in Hawaii, say
808-555-1313,
would be *8085551313*4567*. The PBX parses this string and uses
the
last 4 digits DNIS to route the call from the T1 trunk group to the
proper
internal extension or hunt group.
Do Asterisk and Digium or Sangoma T1/E1 cards know about delimited
ANI
and
DNIS?
--Ron
On Sun, 18 Feb 2007, Eric ManxPower Wieling wrote:
It will do so automatically if it is working. Asterisk will stuff
those
digits into ${EXTEN}, therefore you need an exten =
_XXX,1,Whatever
if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David Ruggles wrote:
Yuan (and Matt),
Thanks for the reply, I'm sorry I kind of vented, I just got
very
frustrated
with trying to configure Asterisk for what (in a proprietary
PBX) is
normally one of the easiest parts of configuration.
With a wink start T1 the DNIS digits are transmitted in-band.
The
Network
goes off hook, the PBX winks (goes off hook for 200ms) and then
the
network
sends DNIS (and ANI if used) as DTMF tones, after the PBX gets
the
tones it
answers the call (goes off hook). So you would tell the PBX to
look
for x
number of digits and then after getting that number of digits it
will answer
the call. I have the Sangoma A101 configured for wink start, but
I
can't
find anything that says how to specify the number DNIS digits to
expect. If
the PBX answers the call instead of just winking, the DTMF tones
will be
transmitted during the call which is what seems to be happening
here.
For more specific information a good overview of the wink start
process can
be found here:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080
1123bb.shtml#topic2a
Can anyone tell me how to configure Asterisk to pickup the DNIS
digits off a
wink start T1?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Yuan
LIU
Sent: Friday, February 16, 2007 5:57 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Does Asterisk support DNIS?
Matt already replied to your other posting of similar
content. I'm
also a
bit confused. Do you mean you have observed that Asterisk is
brought into
the intended context, but start to react to digits in DNIS one
after
another? If so, can you estimate the interval Asterisk stays in
each
extension?
If this is true, it seems to suggest that your provider is
sending
DNIS as a
DTMF string after Asterisk has answered the call. Isn't this a
bit
weird?
What does the card's manual say about DNIS (with wink start)?
Yuan Liu
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Re: [asterisk-users] Summary of Trixbox vs. custom install
You mean compiling raw tar.gz or SRPMS? And where do you download them from? Trixbox site or the original vendors' sites? I just download the tarball from asterisk.org and compile it. Trixbox is not a special version of Asterisk, it is just an easy way to install Asterisk, FreePBX, FOP and a bunch of other packages. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk support DNIS?
Yes, it is complicated. We moved all the telco trunks to PRI last week monday, but the configs for the old settings are still there. We will be moving all the channels going into the Nortel onto a Nortel T-1 card (nor PRI, PRI for Nortel requires a costly feature activation code) soon. /etc/zaptel.conf: loadzone = us defaultzone=us span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs fxsks=1-8 em=9-12 fxsks=13-20 bchan=25-47 dchan=48 fxoks=49-56 em=57-60 fxoks=61-68 /etc/asterisk/zapata.conf: [channels] wink=270 rxwink=270 usecallerid=no echocancel=256 echotraining=900 musiconhold=default toneduration=300 ; Incoming from Telco context=incoming group=1 signalling=fxs_ks channel = 1-8,15-17 ; Incoming from Telco (Property Services) context=temp-propserv-incoming group=0 signalling=fxs_ks channel = 13-14,18-20 ; DID channels from Telco context=incoming group=0 signalling=em_w channel = 9-12 ; Incoming from Channel Bank Corporate callerid=Northpark, Corporate 5558982022 context=toll-access group=2 transfer=yes threewaycalling=yes signalling=fxo_ks channel = 49-56,63-65 ; Incoming from Channel Bank Property Services callerid=Northpark, PropServ 5558980260 context=toll-access group=5 transfer=yes threewaycalling=yes cancallforward=yes signalling=fxo_ks channel = 61-62,66-68 ; DID EM Wink channels to Channel Bank context=INVALID group=3 transfer=no threewaycalling=no cancallforward=no signalling=em_w ; 57 is bad on the nortel channel = 58-60 ; VIVA LE PRI! usecallerid=yes echocancel=no group=4 switchtype=national signalling=pri_cpe context=incoming channel = 25-47 Jason Kim wrote: Would you attach your whole zaptel.conf and zapata.conf? --- C F [EMAIL PROTECTED] wrote: Also check out immediate=no On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: David Ruggles wrote: I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. We had to add this to the /etc/asterisk/zapata.conf to make Asterisk work with the EM Wink start T-1 from our telco. I guess I could paste the settings this time. wink=270 rxwink=270 You might want to play with those settings. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Inbound Problem
HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk PPPD with analog lines
I don't think Asterisk plays a role in this (unless I'm missing your point). A simply script to ping your server room will do. Upon failure, the script could initiate a PPP connection outbound. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dominik Zalewski Sent: Monday, February 19, 2007 6:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk PPPD with analog lines Hi All, Is it possible to use asterisk as a internet link backup callback solution? I mean when my main DSL link is down at my server room I would like to dial to asterisk , then it will call back me and provide a connection to a LAN network. Regards, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Radius users authentication
Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker SIP peer authentication on an external database (RADIUS - LDAP), etc. Although, none of these seems to give me the confidence to implement it in a production environment... What do you people recommend me? Which Asterisk+Radius solution should in your opinion be the best choice? Does Asterisk 1.4 already implement it properly? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Disconnection supervision: what about PBX
From: Stephen Bosch Hi, Trevor: Trevor G. Hammonds wrote: Stephen Bosch wrote: Are BRI circuits what phone companies call digital lines for use with digital sets, such as with digital Centrex? I'm not aware that Telus even offers BRI. Sorry -- BRI is ISDN, not digital Centrex. I'm still not aware that Telus even offers ISDN anymore :) ...and by that I mean ISDN BRI ;) -Stephen- Throughout most of the United States, Digital Centrex or CentrexIS is ISDN as part of a Centrex group. If the circuit is meant for a single device, it would be a BRI. If the circuit is Hi-Cap or meant to be hooked up to a PBX or the like, it would be a PRI. I am not that familiar with Telus, but what Bell is calling Digital Voice service is merely VoIP over one of their DSL connections. While I know that both companies offer Centrex over PRI, I am unsure if either company supports BRI widely anymore. I know BRI service is available, and most of their switches are capable of offering BRI circuits. For example, digital secretarial enhanced key telephone sets are ISDN phones that work via a BRI. In my experience, most telcos in the US and Canada will not tell you about BRI unless you specifically ask. And if you do, they shuffle you off to another department where they may or may not know how to properly provision the circuit. Somehow, all the LECs in North American look at BRI as a data-only service and never really saw the advantages of offering it to voice-only customers. As such, now that 128k (or 144k) is too slow of a data connection for most, BRI has just been passed by. Such a shame... I can still find information pages on BRI on the Telus website (buried, but there); as you point out, though, they refer to data connections only. I am going to give it a try and see what I come up with. There's every possibility they'll offer it but at a ridiculous price, just to discourage adoption enough to let them phase it out. I'll bet that this stuff will disappear when the switching equipment is upgraded. -Stephen- Stephen, I often find that the telcos discourage voice BRI adoption by making it hard for you to obtain the correct information or correct department to order the circuits -- not necessarily by making it overly expensive. I can guarantee that most telcos have no immediate interest in discontinuing BRI (Switched 56, perhaps). However, I cannot tell you how many times I have heard a telco employee say, BRI is for data only. If you can get past the standard business office or residential order center and into the ISDN or complex services group, they will usually be able to help you. In fact, if you have an existing relationship with the telco, the same place where you would order a complex Centrex group or pretty much any T1 would either be able to help you with the BRI order or at least be able to get you to the right place. Take a look at Telus' General Tariff item 485, which covers BRI service: http://about.telus.com/publicpolicy/tariffs/docs2/CRTC180_1/General_2/item48 5.pdf After a brief glance, it looks like Telus charges $91.75 to $107.80 per month (depending on the Rate Band of your exchange) for a 2B+D on a one-year contract. On a five-year contract, that drops to $79.85 to $99.80 per month. Without a five-year commitment, this is quite a bit more than I have seen in Southern California (around US$60/month with the voice feature package). However, California has seemed to be one of the least expensive places for ISDN services. Good luck! Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
I, too, have heard about that best practice of using different channels for different AP's on the same SSID. As far as I can tell, This is standard textbook stuff. Read Cisco press's 'Deploying License Free Wireless Wide-Area Networks' by Jack Unger. it's BS. I don't know who started it, but it has never worked in any of the situations I've encountered. In fact, I know of at least one AP manufacturer (Apple) that has a utility to auto-configure WDS networks, and it auto-configures to use the same channel. That's Using the same channel is bad, because the APs will interfere with each other and your throughput will be reduced. Imagine if you have a total of 2 APs with 10 clients each, the bandwidth will have to be shared amongst the 22 devices. So, if you're able to get 54Mbps on that channel, the net result is everybody gets 54/22 = 2.45Mbps each. Not a very pretty sight. Roaming with multiple APs on the same channel is OK for small set ups. I don't know where one might draw a line between a small and a large setup, but I did one with 15 AP's over a floor of a high-rise. I intially tried the textbook method of different channels, and found the network to be totally useless for either roaming or throughput. I put them all on the same channel and everything was fine. In this case, there were also literally 25 other wireless networks in range with very strong signals (gotta love NYC). I think the moral of the story is that the particular situation will dictate whether or not to use different channels. In a perfect world of evenly distributed AP's with no outside interference, it probably works well to use different channels for adjacent AP's (except for roaming Wifi phones). In a real-world situation with all sorts of 2.4Ghz interference, a single channel may work better. Of course, YMMV. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
The WAP54's have a 'repeater' mode which I've used on occasion. Which is all well and good, but they use WDS which doesn't work with WPA. Not on the WAP54's anyway (I learned the hard way on that one). Some vendors have working solutions: http://expertanswercenter.techtarget.com/eac/knowledgebaseAnswer/0,295199,sid63_gci1104925,00.html - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Summary of Trixbox vs. custom install
From what I've found, if you modify the normal config files, then they will stay until the next time you update your server. When you update your server, they get restored back to the origional, packaged state. It's best to stick to the _custom files as much as possible. John McCollough LAN Network Connections, Inc (603)622-8557 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Sunday, February 18, 2007 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Summary of Trixbox vs. custom install I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? You can just install CentOS with RAID and whatever you want, then use the Trixbox tar package instead of the ISO. Still, why on earth did the Trixbox team didn't leave the option of doing a custom install with the ISO ? 2) How easy it is to find Trixbox SRPMS? Is it possible to compile new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without having to rewrite all the configuration files, changing all paths, all permissions, and so on... You can update Asterisk/zaptel/whatever by just downloading the source and compiling it. My home system was installed with [EMAIL PROTECTED] on version 0.8 (if memory serves me right) and I upgraded Asterisk and Zaptel to version 1.0.10 by downloading and compiling. I know, this is a really old version and I should upgrade, but hey, it is doing everything I need and it is stable (uptime of 315 days). IMHO, Trixbox can me customized alot, but you need to know where and what to modify. I believe that if you know enough about how Asterisk work, you can get around Trixbox limitations. One thing to remember is that the files you can modify are the _custom.conf files. Never touch the _additional.conf files, they will get overwritten next time you click Apply changes in the GUI. The normal base files (sip.conf. iax.conf, etc) can be modified since the GUI doesn't touch them. But I also think that there is nothing that can beat a plain install as far as customization go. YMMV ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Thomas Deillon Envoyé : jeudi, 15. février 2007 11:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option “FAX without T.38(Use G.711 fax)” On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it’s why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Caller ID
I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
I played around the wink and rxwink settings. While increasing rxwink does delay the answer it still sees the DNIS digits individually. I changed the signalling to featd and now I get the following error: WARNING[27630]: chan_zap.c:5661 ss_thread: Got a non-Feature Group D input on channel 1. Assuming EM Wink instead Which I would expect, but the odd thing is that now it's seeing DNIS as a full extension. Going back to the em wink configuration: I found these settings for zapata.conf: prewink: Pre-wink time (default 50ms) preflash:Pre-flash time (default 50ms) wink:Wink time (default 150ms) flash: Flash time (default 750ms) start: Start time (default 1500ms) rxwink: Receiver wink time (default 300ms) rxflash: Receiver flashtime (default 1250ms) debounce:Debounce timing (default 600ms) Can anyone point me to some documentation that explains what these do for em_w signalling? Some of them seem obvious, but don't do what I would expect. With em wink the call answer should progress like this: Network goes off-hook PBX winks (goes off-hook) for 200ms Network sends DNIS as MF/DTMF tones inband PBX goes off-hook and answers. I would assume that wink means the same thing in zapata.conf, so I set it to 200. I also assumed that start meant answer, since there's no other option that seems to match, but changing it around didn't increase or decrease the amount of time it took to answer the call. As I said before changing rxwink did affect the amount of time, but I don't know what it's doing and it doesn't help Asterisk recognize the DNIS. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Caller ID
Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Caller ID
Not sure about an Asterisk only call transfer but in an Asterisk/SER environment the SER server will use the REFER method to perform the transfer. In this case ehe caller ID needs to be the contents of the Refer-By header of the SIP message. Not the contents of EXTEN -Steve Jerry Jones wrote: Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Caller ID
I probably have a screened transfer setup. Is that just a setting somewhere I can easily change? I'm trying to avoid making users press extra keys, like #1 or anything like that. Rob Jerry Jones wrote: Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with T.38
A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, Thomas Deillon [EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Caller ID
Not an asterisk setting. It is how the endpoints perform the transfer. On Feb 19, 2007, at 9:21 AM, Rob Schall wrote: I probably have a screened transfer setup. Is that just a setting somewhere I can easily change? I'm trying to avoid making users press extra keys, like #1 or anything like that. Rob Jerry Jones wrote: Not sure about others, but on Polycoms a blind transfer sends original callerid, screened sends operators callerid On Feb 19, 2007, at 8:55 AM, Rob Schall wrote: I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system (actually the operator). The operator (a real person) answers the call and presses transfer on her polycom 501 phone. I see an incoming call From: Operator. After I pick up her call, she presses transfer one final time to complete the transfer. However, now that the call has been completed, it still shows From: Operator. I need it to show From: 213-555-1234. I tried setting the o setting in Dial, but that didn't seem to fix anything. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with CentOS ztdummy kernel 2.6
did u try modprobe zaptel first ? also check makefile if ztdummy is marked for compilation or not On 19/02/07, Chris Blunt [EMAIL PROTECTED] wrote: Hi List, I am having some trouble with installing the latest version of ztdummy on a CentOS Kernel 2.6 system. I have installed a few Asterisk systems on Slackware Kernel 2.4.x without any issues, unfortunately there is no choice about this distro, or kernel as it has been preinstalled by someone else. And so I am in the dark with an unfamiliar distro and kernel. I am fairly sure the kernel source has been installed. I'm not sure the timer module is installed in the kernel, is it possible to check? If not I think I will need to use ztdummy for definite. Any help with this would be a real life saver. Thanks – Chris From the zaptel-1.2.13 directory I issue the make linux26 command with the following result: make: *** No rule to make target `linux26'. Stop. Just issuing the make command does seem to work and concludes with: make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Make install outputs the following: make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 HOTPLUG_FIRMWARE=yes modules make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ install -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ install -m 644 wctc4xxp/*.bin /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ install -m 644 wct4xxp/*.ima /lib/firmware; \ install -m 644 wctc4xxp/*.bin /lib/firmware; \ fi Installed firmware install -D -m 755 ztcfg /sbin/ztcfg if [ -f sethdlc-new ]; then \ install -D -m 755 sethdlc-new /sbin/sethdlc; \ elif [ -f sethdlc ]; then \ install -D -m 755 sethdlc /sbin/sethdlc ; \ fi if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi for x in zaptel.ko tor2.ko torisa.ko wcusb.ko wcfxo.ko wctdm.ko wctdm24xxp.ko ztdynamic.ko ztd-eth.ko wct1xxp.ko wcte11xp.ko pciradio.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/extra/$x ; \ done; \ make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' INSTALL /usr/src/zaptel-1.2.13/pciradio.ko INSTALL /usr/src/zaptel-1.2.13/tor2.ko INSTALL /usr/src/zaptel-1.2.13/torisa.ko INSTALL /usr/src/zaptel-1.2.13/wcfxo.ko INSTALL /usr/src/zaptel-1.2.13/wct1xxp.ko INSTALL /usr/src/zaptel-1.2.13/wct4xxp/wct4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctc4xxp/wctc4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctdm.ko INSTALL /usr/src/zaptel-1.2.13/wctdm24xxp.ko INSTALL /usr/src/zaptel-1.2.13/wcte11xp.ko INSTALL /usr/src/zaptel-1.2.13/wcusb.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxo.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxs.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp_usb.ko INSTALL /usr/src/zaptel-1.2.13/zaptel.ko INSTALL /usr/src/zaptel-1.2.13/ztd-eth.ko INSTALL /usr/src/zaptel-1.2.13/ztd-loc.ko INSTALL /usr/src/zaptel-1.2.13/ztdummy.ko INSTALL /usr/src/zaptel-1.2.13/ztdynamic.ko INSTALL /usr/src/zaptel-1.2.13/zttranscode.ko make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' if ! [ -f wcfxsusb.o ]; then \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxsusb.o; \ fi; \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxs.o install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 [ `id -u` = 0 ] /sbin/ldconfig || : rm -f /usr/lib/libtonezone.so ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then /sbin/restorecon -v /usr/lib/libtonezone.so; fi install -D -m 644 zaptel.h /usr/include/linux/zaptel.h install -D -m 644 torisa.h /usr/include/linux/torisa.h install -D -m 644 tonezone.h /usr/include/tonezone.h install -m 644 doc/ztcfg.8 /usr/share/man/man8 install -m 644 doc/zttool.8 /usr/share/man/man8 [ `id -u` = 0 ] /sbin/depmod -a 2.6.9-42.0.3.ELsmp || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample/etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc wct4xxp wcfxs wctdm8xxp wct2xxp Building /etc/modprobe.conf... Once it is installed I run: modprobe ztdummy with the following result. FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy -- Chris Blunt Entropy IT Ltd ___ --Bandwidth and Colocation provided
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
- Original Message - From: Nic Bellamy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 16, 2007 12:56 AM Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Wireless wrote: Thanks Nic, I have bought a couple of HPEC channel licences from Digium and been trying to get them working, all seems fine until I get to 9 and 10 of this doc ftp://ftp.digium.com/pub/telephony/hpec/README - at which point Asterisk is not running and I've issued a: wanrouter start command and all looks good. 9 says type [EMAIL PROTECTED] ~]# modprobe zaptel which returns nothing... when I run 10 At this point, if you run dmesg, do you find the following in your kernel log? Digium High-Performance Echo Canceller, version 8.20 Optimized for i386 CPU architecture Coypright (C) 2006 Digium, Inc. and Adaptive Digital Technologies, Inc. This module is supplied under a commercial license granted by Digium, Inc. Please see the full license text supplied by the accompanying register utility, or ask for a copy from Digium. If not, you've probably not got Zaptel built with HPEC properly. [EMAIL PROTECTED] ~]# ./zaphpec_enable I get - No valid licenses for HPEC found. If anyone can shed I bit of light on how to register my licence I'd be very greatful, I've checked in /var/lib/digium/licenses and there is a licence there. Hmm... not run into this myself - after registering my key, it worked first pop for me, giving the following output: # ./zaphpec_enable Digium High-Performance Echo Canceller Enabler Copyright (C) 2006, Digium, Inc. Version 1.0.0 Use the '-l' option to see license information for software included in this program. Found key 'HPEC-' for 4 channels. Found valid HPEC licenses for 4 channels. Successfully enabled 4 channels. After this, the follow line is spat out by the kernel: hpec_license_check: License granted for 4 channels Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ I'm truely stuck now, I cannot get HPEC to register with my Sangoma A200 card. I'm using Asterisk 1.2.15 Zaptel 1.2.13 Wanpipe drivers / util 2.3.4-7 I'm just not seeing any mention of HPEC in dmesg and I have tried different versions of the HPEC i386, i586, i686 and pentium3m the physical proc is a P3 650Mhz running CentOS 4.4 (Trixbox 2) I've rebuilt this box over the weekend from a fully patched CentOS 4.4 (yum update) as the hard drive failed! when I run ./register all seems ok then when I run ./zaphpec_enable it reports: No valid licenses for HPEC found. Any suggestions as to how I can debug what is not happening much appreciated Thanks Harvey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP resigtrations and OpenSer
I have an ITSP provider that will only deliver calls using SIP registrations (would prefer delivery to static IAX or SIP url, but hey), periodically their servers don't respond to a renew request, and when this happens the sip stack in asterisk (1.4.0) stops working until either a SIP reload is issued (or sometimes a restart now). I'm wondering if this can be solved by installing OpenSER, and using that to register with the remote provider and redirect to asterisk through a single static sip trunk. Are there any other solutions that I haven't thought of. TIA for any help with this matter. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx with ASTERISK 1.4
Thank You all, thank you very much ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE110P
Hello, i've installed trixbox with TE110P TDM400B, but no led is ON in the TE110P, i don't know why even if the 4 leds of My TDM are greens any explaination Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 80
-- Message: 2 Date: Mon, 19 Feb 2007 13:03:05 +0200 From: Eugeniy Khvastunov [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk and a modem pool. To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=KOI8-U; format=flowed Asterisk and a modem pool. I have a small modem pool, after its carry for an asterisk modems have ceased will incorporate or incorporate, but for the speed 9600. I use Tormenta 2b2. Prompt the decision. -- Has started an asterisk in a test mode (in an asterisk two streams - two operators, from it one stream on mini ats) and find a following raker: 1) at 8 and more passing calls through * on mini ats quality of a voice (, places, letters in words noticeably worsens Places an echo...) 2) on mini ats is modems begin a small modem pool droop call I so I understand owing to 1) Read that such can be at insufficient speed of processing, but loading ? does not jump more than 8 %. I Use Asterisk 1.2.11 built by root voip on a i686 running Linux on 2006-10-03 12:15:54 UTC on Gentoo Help why so it turns out? Business in adjustments or in the card? Here mine zapata.conf: [trunkgroups] [channels] callprogress=yes usecallerid=yes hidecallerid=no restrictcid=no switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown busydetect=yes busycount=8 transfer=yes callwaiting=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes progzone=ru rxgain=8.2 txgain=1.0 context=it group = 1 context=internal signalling=pri_cpe channel = 1-15 channel = 17-31 group = 2 context=velton signalling=pri_cpe channel = 32-46 channel = 48-62 group = 3 context=datagroup signalling=pri_cpe channel = 63-77 channel = 79-93 group = 4 context=gsmgate signalling=pri_cpe channel = 94-108 channel = 110-124 -- wbr. Eugeniy Khvastunov aka FreeMan *** http://unlimite.org.ua ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 19 Feb 2007, at 15:01, Noah Miller wrote: The WAP54's have a 'repeater' mode which I've used on occasion. Which is all well and good, but they use WDS which doesn't work with WPA. Not on the WAP54's anyway (I learned the hard way on that one). Some vendors have working solutions: Apple Airports do WDS and WPA/WPA2 just fine. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF2dEBRAx5nvEhZLIRAnYYAJ0W9Bc+yredI/++EQgUPwvSDBLtXACgl5rp f/tzrwHxlf6Me8MVx1H7l4k= =Wz5J -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with CentOS ztdummy kernel 2.6
On Mon, Feb 19, 2007 at 09:08:41PM +0530, Mail list wrote: Once it is installed I run: modprobe ztdummy with the following result. did u try modprobe zaptel first ? also check makefile if ztdummy is marked for compilation or not No. the module iwas not found. 'modinfo ztdummy' won't show it. maybe it is just that depmod wasn't run? run 'depmod' and try modinfo ztdummy again. Also: is the module zaptel availble? modionfo zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rajnish Jain Envoyé : lundi, 19. février 2007 16:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, Thomas Deillon [EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Transfer Caller ID
RS == Rob Schall [EMAIL PROTECTED] writes: RS Example: John calls in from the outside using (213-555-1234) and RS he calls into the asterisk system (actually the operator). The RS operator (a real person) answers the call and presses transfer on RS her polycom 501 phone. I see an incoming call From: Operator. Yes, that is the only incoming call, and it is from Operator. There is no association with the other call, so it cannot show anything else. RS After I pick up her call, she presses transfer one final time to RS complete the transfer. However, now that the call has been RS completed, it still shows From: Operator. I need it to show RS From: 213-555-1234. You can't. The call is still the same, and you can't change caller ID in the middle of a call. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer with snom phones
I have setup an asterisk based phone system using snom-320 (SIP based) phones. I would like to change what seems to be the default procedure for an attended call transfer. Right now, the phone user places the call on hold, calls the extension using a extension button on the phone, speaks with the call recipient , and presses transfer to transfer the held call. The users would like to press transfer, call the extension using a extension button on the phone, speak to the recipient, then hangup to complete the call. Can you give me a suggestion as to how to do this. Thank you, Michael Boers Michael Scott Technology ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with CentOS ztdummy kernel 2.6
is it a spinlock problem? try http://www.trixbox.org/modules/newbb/viewtopic.php?viewmode=flattopic_id=1626forum=2 - Original Message - From: Mail list To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 19, 2007 3:38 PM Subject: Re: [asterisk-users] Problems with CentOS ztdummy kernel 2.6 did u try modprobe zaptel first ? also check makefile if ztdummy is marked for compilation or not On 19/02/07, Chris Blunt [EMAIL PROTECTED] wrote: Hi List, I am having some trouble with installing the latest version of ztdummy on a CentOS Kernel 2.6 system. I have installed a few Asterisk systems on Slackware Kernel 2.4.x without any issues, unfortunately there is no choice about this distro, or kernel as it has been preinstalled by someone else. And so I am in the dark with an unfamiliar distro and kernel. I am fairly sure the kernel source has been installed. I'm not sure the timer module is installed in the kernel, is it possible to check? If not I think I will need to use ztdummy for definite. Any help with this would be a real life saver. Thanks – Chris From the zaptel-1.2.13 directory I issue the make linux26 command with the following result: make: *** No rule to make target `linux26'. Stop. Just issuing the make command does seem to work and concludes with: make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Make install outputs the following: make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 HOTPLUG_FIRMWARE=yes modules make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' Building modules, stage 2. MODPOST make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' build_tools/genudevrules /etc/udev/rules.d/zaptel.rules if [ -d /usr/lib/hotplug/firmware ]; then \ install -m 644 wct4xxp/*.ima /usr/lib/hotplug/firmware; \ install -m 644 wctc4xxp/*.bin /usr/lib/hotplug/firmware; \ fi if [ -d /lib/firmware ]; then \ install -m 644 wct4xxp/*.ima /lib/firmware; \ install -m 644 wctc4xxp/*.bin /lib/firmware; \ fi Installed firmware install -D -m 755 ztcfg /sbin/ztcfg if [ -f sethdlc-new ]; then \ install -D -m 755 sethdlc-new /sbin/sethdlc; \ elif [ -f sethdlc ]; then \ install -D -m 755 sethdlc /sbin/sethdlc ; \ fi if [ -f zttool ]; then install -D -m 755 zttool /sbin/zttool; fi for x in zaptel.ko tor2.ko torisa.ko wcusb.ko wcfxo.ko wctdm.ko wctdm24xxp.ko ztdynamic.ko ztd-eth.ko wct1xxp.ko wcte11xp.ko pciradio.ko ztd-loc.ko ztdummy.ko zttranscode.ko; do \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/extra/$x ; \ done; \ make -C /usr/src/linux SUBDIRS=/usr/src/zaptel-1.2.13 INSTALL_MOD_PATH= INSTALL_MOD_DIR=misc modules_install; make[1]: Entering directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' INSTALL /usr/src/zaptel-1.2.13/pciradio.ko INSTALL /usr/src/zaptel-1.2.13/tor2.ko INSTALL /usr/src/zaptel-1.2.13/torisa.ko INSTALL /usr/src/zaptel-1.2.13/wcfxo.ko INSTALL /usr/src/zaptel-1.2.13/wct1xxp.ko INSTALL /usr/src/zaptel-1.2.13/wct4xxp/wct4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctc4xxp/wctc4xxp.ko INSTALL /usr/src/zaptel-1.2.13/wctdm.ko INSTALL /usr/src/zaptel-1.2.13/wctdm24xxp.ko INSTALL /usr/src/zaptel-1.2.13/wcte11xp.ko INSTALL /usr/src/zaptel-1.2.13/wcusb.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxo.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpd_fxs.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp.ko INSTALL /usr/src/zaptel-1.2.13/xpp/xpp_usb.ko INSTALL /usr/src/zaptel-1.2.13/zaptel.ko INSTALL /usr/src/zaptel-1.2.13/ztd-eth.ko INSTALL /usr/src/zaptel-1.2.13/ztd-loc.ko INSTALL /usr/src/zaptel-1.2.13/ztdummy.ko INSTALL /usr/src/zaptel-1.2.13/ztdynamic.ko INSTALL /usr/src/zaptel-1.2.13/zttranscode.ko make[1]: Leaving directory `/usr/src/kernels/2.6.9-42.0.3.EL-i686' if ! [ -f wcfxsusb.o ]; then \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxsusb.o; \ fi; \ rm -f /lib/modules/2.6.9-42.0.3.ELsmp/misc/wcfxs.o install -D -m 755 libtonezone.so /usr/lib/libtonezone.so.1.0 [ `id -u` = 0 ] /sbin/ldconfig || : rm -f /usr/lib/libtonezone.so ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so.1 ln -sf libtonezone.so.1.0 \ /usr/lib/libtonezone.so if [ -x /usr/sbin/sestatus ] (/usr/sbin/sestatus | grep SELinux status: | grep -q enabled) ; then /sbin/restorecon -v /usr/lib/libtonezone.so; fi install -D -m 644 zaptel.h /usr/include/linux/zaptel.h install -D -m 644 torisa.h /usr/include/linux/torisa.h install -D -m 644 tonezone.h /usr/include/tonezone.h install -m 644 doc/ztcfg.8
Re: [asterisk-users] Asterisk Inbound Problem
Without seeing your config files my guess would be that this is something to do with a bad codec negotiation. I'd bet that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7941/7961 w/Firmware 8.2.1 and NAT
Hi all, I have a bunch of Cisco 7941's with SIP firmware 8.2.1 that work great on a LAN with Asterisk 1.2.x but fail miserably when trying to use them, through NAT, with a public Asterisk server. We have a bunch of Polycoms and Aastra phones that work great with NAT so I'm very familiar with NAT and Asterisk but not so much familiar with the Cisco phones although we did get them to work great on our LAN and local Asterisk 1.2.x box. I've heard mixed opinions, some say it will never work, some say it should work so I've resorted to the mailing list for help. So, I'm looking for an example sip.conf and SEPmac.cnf.xml file for either 7941's or 7961's inside a NAT network going to a public Asterisk 1.2.x server. Hopefully, for phone firmware revision 8.2 or close. Any help would be greatly appreciated! - Bill *** This message is confidential and intended only for the listed recipient(s). Any email sent to the originator is subject to monitoring and review. No liability is assumed regarding the content of this message or any replies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Attended Transfer with snom phones
MB == Michael Boers [EMAIL PROTECTED] writes: MB I have setup an asterisk based phone system using snom-320 (SIP MB based) phones. MB I would like to change what seems to be the default procedure for MB an attended call transfer. Right now, the phone user places the MB call on hold, calls the extension using a extension button on MB the phone, speaks with the call recipient , and presses transfer MB to transfer the held call. MB The users would like to press transfer, call the extension using a MB extension button on the phone, speak to the recipient, then MB hangup to complete the call. Reprogram the transfer button to be a hold button, and make sure transfer-on-onhook (or is it transfer-on-hangup?) is turned on in the phone. I think that should work, I can check when I'm at work tomorrow. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UTStarcom F1000 - WLAN connection unreliable
Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available (although it is, with the other phone or a laptop). It will only re-find the WLAN after either powering down the phone, or going into the WLAN settings menu, down to any setting, OK'ing that and activating that WLAN setting. I used any of the profiles 1 to 4 in the meantime, all the same results. I tried changing from WPA to WEP-128 to unencrypted WLAN, IP via DHCP versus static IP, DNS via DHCP (while IP came from DHCP) versus static DNS server, registering to a domain name versus registering to the appropriate IP address - to no avail. I had both phones turned on at times, or only one, that would not make a difference. This occurs with both phones, and on Accesspoints from Buffalo(OpenWRT), Fon (LaFonera), AVM (FritzBoxFon 7050), and T-Com (Eumex something). I did not cross-test all possible combinations - that would be a lot - but quite some. Does anyone know of those problems, and possibly have a solution? Or just a good idea? Is there a known reliable setup? Would anyone care to post what makes his asterisk work with the F1000 (WLAN settings, and sip.conf settings, just to go sure?) Would chances of a working setup increase with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open CallerID Database?
Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good 4 Port PSTN Gateway
Hi All I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura 2000) and a CG-410 Gateway to connect the the two PSTN lines that I have. I have a odd hassle that for no apparent reason, the calls will quite working. but quite I mean the phones will ring but their is no voice packets. There is NO NAT issues as it is all LAN or internal. The Gateway IP Phones and PC are connected to a switch. The only questionable traffic is I run Citrix out the WAN to an outside provider. Yet the phones quite working is very very random no predictable pattern When I reset the phones and the gateway all is well again. the Asterisk PBX manages the calls. Any advice will be welcome Thanks all Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Open CallerID Database?
Hey Shane, The basis of my idea was that it would be user-moderated/generated. A 'owner/operator' of a number, would submit verify their phone number, enter their caller id, and basically be done with it. The logistics of it I don't really think would be that complicated. If a listing needs to be updated they basically go through the same process. Right now, we're using a commonly available script (I can't remember the link off hand) that uses Google, 411.com, etc, to do a lookup and although it works pretty good, it is horribly inaccurate the majority of the time. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Monday, February 19, 2007 12:46 PM To: Robert Norton - SophMedia LLC Subject: Re: [asterisk-users] Open CallerID Database? Robert On the surface, I don't see how you could a db with a very good hit rate without paying for the data. There are thousands and thousdands of database updates every day. Perhaps I am missing your intent here. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. --Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium TE110P
Hello, i've installed trixbox with TE110P TDM400B, but no led is ON in the TE110P, i don't know why even if the 4 leds of My TDM are greens any explaination Thank You No LEDs on TE110P (and similar cards) can sometimes mean that the Zaptel driver isn't running. Can you run zttool and see anything happening, even red or blue alarms? Also, have you been able to confirm that your drivers are even loaded? Do: lsmod and make sure you have your drivers: zaptel, wcte11xp I don't personally have a TE110P so I can't offer you any advice specific to this card... Let us know what happens. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting Caller-ID / Point Codes
Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is currently asking questions about the LOA, and I am concerned they may not sign it. What do other people here know about this procedure. Have any of you signed up with a company to allow you to set the Caller-ID-Name? If so, was an LOA required? Did your CLEC sign it? Who do you all work with? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. A public database managed by users would be of little value unless there are means to verify the data. If people trusted it, outbound telemarketers might try to put friendly CNAM values in. Robert Norton - SophMedia LLC wrote: Hey Guys, I’m curious if there’s an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks – Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. YES! Would creating a public database, managed by users be worthwhile to anyone? I'm not sure the technical issues will be as easy to work out as one would hope. When creating such a system, care must be taken to keep the information accurate and up-to-date. And where would you get the information from in the first place? Thanks – Any input is greatly appreciated. What I would like to see is a distributed system that allows for updates to be rsync'd in, so that those of us who keep our servers off the Internet can move it through a QA process and then push the update through. Some type of a mirror system, where the packages can be updated from time to time (like daily). -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com http://www.xstreamhost.com/ - Web Hosting http://www.SophMedia.com http://www.sophmedia.com/ - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
Am Montag, den 19.02.2007, 12:39 -0700 schrieb Robert Norton - SophMedia LLC: Hey Guys, I’m curious if there’s an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Sounds like a great thing to have, but I doubt this is possible without violating existing laws. At least in Europe there are some quite restricting rules concerning the storage and transmission of people-related data. Without explicit permission you might not be allowed to transmit people's unlisted phone numbers, for example. Take care, An(not a lawyer)selm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Open CallerID Database?
Our CNAM provider claims to have more than 196 million entries. I just don't think you could reliably maintain that in this format. Let's say I'm a CLEC and I have 40,000 numbers. I want to update that in one place (my SCP, probably). I wouldn't also want to update another database through another method. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Shane, The basis of my idea was that it would be user-moderated/generated. A 'owner/operator' of a number, would submit verify their phone number, enter their caller id, and basically be done with it. The logistics of it I don't really think would be that complicated. If a listing needs to be updated they basically go through the same process. Right now, we're using a commonly available script (I can't remember the link off hand) that uses Google, 411.com, etc, to do a lookup and although it works pretty good, it is horribly inaccurate the majority of the time. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Monday, February 19, 2007 12:46 PM To: Robert Norton - SophMedia LLC Subject: Re: [asterisk-users] Open CallerID Database? Robert On the surface, I don't see how you could a db with a very good hit rate without paying for the data. There are thousands and thousdands of database updates every day. Perhaps I am missing your intent here. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. --Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: looks good. 9 says type [EMAIL PROTECTED] ~]# modprobe zaptel which returns nothing... when I run 10 At this point, if you run dmesg, do you find the following in your kernel log? Digium High-Performance Echo Canceller, version 8.20 Optimized for i386 CPU architecture Coypright (C) 2006 Digium, Inc. and Adaptive Digital Technologies, Inc. This module is supplied under a commercial license granted by Digium, Inc. Please see the full license text supplied by the accompanying register utility, or ask for a copy from Digium. If not, you've probably not got Zaptel built with HPEC properly. I'm truely stuck now, I cannot get HPEC to register with my Sangoma A200 card. I'm using Asterisk 1.2.15 Zaptel 1.2.13 Wanpipe drivers / util 2.3.4-7 I'm just not seeing any mention of HPEC in dmesg and I have tried different versions of the HPEC i386, i586, i686 and pentium3m the physical proc is a P3 650Mhz running CentOS 4.4 (Trixbox 2) I've rebuilt this box over the weekend from a fully patched CentOS 4.4 (yum update) as the hard drive failed! when I run ./register all seems ok then when I run ./zaphpec_enable it reports: No valid licenses for HPEC found. Any suggestions as to how I can debug what is not happening much appreciated Before building Zaptel, you are grabbing the correct version of hpec_x86_32.o_shipped for your CPU and putting it in zaptel-1.2.13/hpec/ right? It sounds to me like you've either not done that correctly, or something with the Sangoma build process is stopping the HPEC build working. After building zaptel, run strings zaptel.ko | grep 'High-Performance Echo Canceller' and see if you get a line like: Digium High-Performance Echo Canceller, version %s If not, you're going to need to dig into the way your Zaptel is being built to see why the HPEC module is not being included. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Open CallerID Database?
You MUST account for fraud, as well. Perhaps proving you own the number, as in the LNP process, by providing the cover page of the bill... _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion Sent: Monday, February 19, 2007 3:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open CallerID Database? On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. YES! Would creating a public database, managed by users be worthwhile to anyone? I'm not sure the technical issues will be as easy to work out as one would hope. When creating such a system, care must be taken to keep the information accurate and up-to-date. And where would you get the information from in the first place? Thanks - Any input is greatly appreciated. What I would like to see is a distributed system that allows for updates to be rsync'd in, so that those of us who keep our servers off the Internet can move it through a QA process and then push the update through. Some type of a mirror system, where the packages can be updated from time to time (like daily). -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com http://www.xstreamhost.com/ - Web Hosting http://www.SophMedia.com http://www.sophmedia.com/ - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
It seems like an interesting idea, but if this would be a public user updated 411, who would ensure that was more up to date than 411. If the numbers are off from 411, then the phone provider isn't keeping the records properly. A customer should be notifying the phone company when they are moving, etc, so when people want to find their business via 411, it should resolve correctly. If a customer does make that change, unless they, or another user who knows of their change, updates the public user run DB, then it will be outdated in no time. Seems like you would want some type of hybrid. Have a weekly check of all the submitted numbers and look for any changes in 411. If that updates, then the user's information would be updated, and the person who submitted it would be contacted. Although I still think 411 would remain more up to date, it probably would be better than just relying on a group of users to keep businesses information up to date. Rob Anselm Martin Hoffmeister wrote: Am Montag, den 19.02.2007, 12:39 -0700 schrieb Robert Norton - SophMedia LLC: Hey Guys, I’m curious if there’s an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Sounds like a great thing to have, but I doubt this is possible without violating existing laws. At least in Europe there are some quite restricting rules concerning the storage and transmission of people-related data. Without explicit permission you might not be allowed to transmit people's unlisted phone numbers, for example. Take care, An(not a lawyer)selm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
At least the current system involves some qualification(CLEC status) and some policing(regulators can revoke CLEC status). Why create a system where setting your CNAM requires about as much validation as registering a domain name? You need to consider that most of the free data from sources like 411.com can't be legally used in such a database, even for verification purposes. Read all the fine print before you get too excited. I will agree that the traditional system is not flawless. For example, the ILEC did not check my articles of incorporation to see if the business name was valid. However, scammers who go so far as to rent space and install business phone service are a lot easier to trace than those who buy a SIP DID using a prepaid Visa gift card. David Gomillion wrote: On 2/19/07, *Robert Norton - SophMedia LLC* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. YES! Would creating a public database, managed by users be worthwhile to anyone? I'm not sure the technical issues will be as easy to work out as one would hope. When creating such a system, care must be taken to keep the information accurate and up-to-date. And where would you get the information from in the first place? Thanks – Any input is greatly appreciated. What I would like to see is a distributed system that allows for updates to be rsync'd in, so that those of us who keep our servers off the Internet can move it through a QA process and then push the update through. Some type of a mirror system, where the packages can be updated from time to time (like daily). -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com http://www.xstreamhost.com/ - Web Hosting http://www.SophMedia.com http://www.sophmedia.com/ - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7912 phones loosing registration
Jerry, Thats a really interesting question... I have (aparrently) identical 7912 phones and some stay registered correctly for every while others drop out and display the X. I've never spent the time (yet) to investigate it further, but as they are all running the same version of SIP firmware I guess (!!) it might be something I've done different in the configuration... either of the phone or Asterisk... Mike - Original Message - From: Jerry Geis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 15, 2007 6:57 PM Subject: [asterisk-users] 7912 phones loosing registration I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7912 phones loosing registration
Michael J. Tubby G8TIC wrote: Thats a really interesting question... I have (aparrently) identical 7912 phones and some stay registered correctly for every while others drop out and display the X. If you guys are using DHCP for your phones, make sure that your lease time isn't too short. It was causing issues for me. Set the time for 30 days and people stopped complaining. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel and zaptel versions
Hello, Can anyone recommend the 'best' kernel and zaptel versions to use with asterisk? we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can recommend specific software versions that have been proven to be stable and reliable? We're running trixbox 1.2.3 in case anyone has similar issues that they might know how to solve Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nnpb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
- Original Message - From: Nic Bellamy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 19, 2007 8:21 PM Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Wireless wrote: looks good. 9 says type [EMAIL PROTECTED] ~]# modprobe zaptel which returns nothing... when I run 10 At this point, if you run dmesg, do you find the following in your kernel log? Digium High-Performance Echo Canceller, version 8.20 Optimized for i386 CPU architecture Coypright (C) 2006 Digium, Inc. and Adaptive Digital Technologies, Inc. This module is supplied under a commercial license granted by Digium, Inc. Please see the full license text supplied by the accompanying register utility, or ask for a copy from Digium. If not, you've probably not got Zaptel built with HPEC properly. I'm truely stuck now, I cannot get HPEC to register with my Sangoma A200 card. I'm using Asterisk 1.2.15 Zaptel 1.2.13 Wanpipe drivers / util 2.3.4-7 I'm just not seeing any mention of HPEC in dmesg and I have tried different versions of the HPEC i386, i586, i686 and pentium3m the physical proc is a P3 650Mhz running CentOS 4.4 (Trixbox 2) I've rebuilt this box over the weekend from a fully patched CentOS 4.4 (yum update) as the hard drive failed! when I run ./register all seems ok then when I run ./zaphpec_enable it reports: No valid licenses for HPEC found. Any suggestions as to how I can debug what is not happening much appreciated Before building Zaptel, you are grabbing the correct version of hpec_x86_32.o_shipped for your CPU and putting it in zaptel-1.2.13/hpec/ right? It sounds to me like you've either not done that correctly, or something with the Sangoma build process is stopping the HPEC build working. After building zaptel, run strings zaptel.ko | grep 'High-Performance Echo Canceller' and see if you get a line like: Digium High-Performance Echo Canceller, version %s If not, you're going to need to dig into the way your Zaptel is being built to see why the HPEC module is not being included. Cheers, Nic. Hi Nic Thanks for that, it does indeed show Digium High-Performance Echo Canceller, version %s, I've emailed Digium support but not sure if they will help me as I'm using the Sangoma card - here hoping :) Harvey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto load of zap drivers
My understanding is that with Asterisk 1.2.x issuing the command of make install-udev allowed the drivers to be loaded upon the server boot. Doing this with version 1.4 does not seem to work. Using menuselect I selected zaptel and ztdummy. Should I also be selecting something else for the drivers to load at start up? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel and zaptel versions
Why install Centos -- its really old? Check with whoever is supplying your telephony hardware and see what kernel versions are needed or will work with that hardware. on Monday 02/19/2007 mail-lists([EMAIL PROTECTED]) wrote Hello, Can anyone recommend the 'best' kernel and zaptel versions to use with asterisk? we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can recommend specific software versions that have been proven to be stable and reliable? We're running trixbox 1.2.3 in case anyone has similar issues that they might know how to solve Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
First my two cents. I dont think creating a system to store the info is a good idea mostly because of the reasons mentioned in this thread. An easy way to do free CNAM lookup, keep one pots line around with caller id service on it for each non cached number that comes in make a phone call to the pots line with caller id set to the number of the unknown name, the caller id service on the pots line should reveal the name after the first ring. cache that and reuse it as needed. On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good 4 Port PSTN Gateway
I use the mediatrix 1204 although not easy to configure it does an excellent job On 2/19/07, Barry Fawthrop [EMAIL PROTECTED] wrote: Hi All I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura 2000) and a CG-410 Gateway to connect the the two PSTN lines that I have. I have a odd hassle that for no apparent reason, the calls will quite working. but quite I mean the phones will ring but their is no voice packets. There is NO NAT issues as it is all LAN or internal. The Gateway IP Phones and PC are connected to a switch. The only questionable traffic is I run Citrix out the WAN to an outside provider. Yet the phones quite working is very very random no predictable pattern When I reset the phones and the gateway all is well again. the Asterisk PBX manages the calls. Any advice will be welcome Thanks all Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP interface status and calllimit
There is an issue when using call-limit for a SIP interface in sip.conf. The call count does not properly reset when some calls end. The problem happens regardless of which side of the connection ends the call. It happens on all calls including calls from SIP interface to SIP interface (with no reinvite) within the same Asterisk server. I have not been able to determine a definite pattern. I can call from one interface to another 50 times before it happens and sometimes it happens after only 2 calls. We have to enable call-limit for our customer service queue agents so that the ringinuse option in queues.conf will work properly. Has anyone else seen this issue? Any ideas? Thanks, James Fromm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
C F, your method is technically feasible. I thought of it over a year ago but never posted it because it would definitely be considered an abuse by most providers. If a lot of people start using it, I expect we would start seeing TOS for PSTN and voip alike to start changing. Of course the best thing that could happen is for CNAM database access costs to become a lot more reasonable. My pots line is with a small rural ILEC who depends on verisign for CNAM matters. My guess is that they pay a high rate and a large number of unanswered calls from different numbers would be noticed. I get the impression that the CNAM system has an air of gangsterism and racketeering about it. C F wrote: First my two cents. I dont think creating a system to store the info is a good idea mostly because of the reasons mentioned in this thread. An easy way to do free CNAM lookup, keep one pots line around with caller id service on it for each non cached number that comes in make a phone call to the pots line with caller id set to the number of the unknown name, the caller id service on the pots line should reveal the name after the first ring. cache that and reuse it as needed. On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
Does google really have the true CNAM database? When I enter my number, I get a search result for my business listing at yellowpages.com Are you referring to something available in a google area other than the search engine? Brad Templeton wrote: On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nnpb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
I doubt it's CNAM since it has old an outdated listings. On 2/19/07, Paul [EMAIL PROTECTED] wrote: Does google really have the true CNAM database? When I enter my number, I get a search result for my business listing at yellowpages.com Are you referring to something available in a google area other than the search engine? Brad Templeton wrote: On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nnpb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
My better guess is, that since Google is a web search engine all their results are based on web gatherings, which is where Google is taking the number resluts from. On 2/19/07, Paul [EMAIL PROTECTED] wrote: Does google really have the true CNAM database? When I enter my number, I get a search result for my business listing at yellowpages.com Are you referring to something available in a google area other than the search engine? Brad Templeton wrote: On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote: I think terms of service for most CNAM providers prohibits sharing the data and limits the amount of time it can be cached for your own reuse. I don't know why they manage to get this level of control over the cnam database so that they can charge a penny per lookup as well as monthly fees. Does anybody know how this happens? Clearly some people buy the database at a good price. Google for example has it, and there are asterisk hacks to do google lookup (if you query a 10 digit phone number in google, you'll get not just name but address etc.) Perhaps they are just paying. One way to build a free database would be to simply have people share the results of all sorts of searches. People who pay for CNAM as end users, for example, have signed no contract to not share the data. So they could, if trusted, forward those records to be stored in the shared database. People who don't could take any number they get, and if it's not in the shared database already, do a google query, and if that gives a result, store that in the shared database. (Also store negative results with a timestamp so that you know that the google lookup provides no info.) http://www.google.com/search?q=nnpb=r Eventually you would get a pretty good database, perhaps one big enough that CLECs start wanting to update it directly? Now there may still need to be something to pay for all of this, but the fees could be much lower. Charge fees for the latest copy or real time query but just have the regular database out there for download and local lookup. Or perhaps just use the google api? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.4 svn fails to compile (xbus-core.c)
This compile error started happening about 2 weeks ago with zaptel. /mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’: /mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘u’ make[4]: *** [/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.o] Error 1 BTW - Is there a separate zaptel mailing list? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: FW: [asterisk-users] Problem Transferring Direct to Voicemail
I have filed a bug report on this issue since it does work in 1.2.9.1 but not in 1.4.0 and I tried the latest SVN and it still doesn't work. If anyone else can try to reproduce this and see if they get the same error it would be appreciated. The dialplan should have _*40XX,1,VoicemailMain($(EXTEN},u) or match your voicemail setup From: [EMAIL PROTECTED] on behalf of Stephen Bosch Sent: Fri 2/16/2007 6:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail Savoy, Kevin - Williston, ND wrote: Well thanks to those who did reply. I guess I'll have to live with it until somehow it gets fixed. The reason I upgraded to 1.4 is that there were three or four other issues I had that this fixed. Going back just isn't really an option since those issues were bigger then this one. Guess we'll live with it for now. I strongly encourage you to file a bug, as the developers need feedback to make improvements on 1.4. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel and zaptel versions
we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can recommend specific software versions that have been proven to be stable and reliable? What kind of voice connections are you using? VoIP? PSTN? What is your hardware? You can probably solve quality issues without doing a complete reinstall, and you should be able to get reliable results with most all kernel versions and asterisk versions. There are some exceptions, but versions of asterisk and the linux kernel that come with trixbox have generally been tested to work before trixbox gets released. If your connections are VoIP, the first area to look at for quality is network jitter/congestion/drops. If your connections are analog PSTN, can you get reliable connections without asterisk in the picture? Also, if PSTN, is your voice card sharing interrupts with anything else? Why install Centos -- its really old? Does that make it less good? I've found CentOS to be exceptionally stable, which is my most important criteria. It has too many services running by default, but that's an easy problem to solve. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
Instead of forwarding to IAX softphone if I'll play some music same thing is happening in this case also. On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote: Without seeing your config files my guess would be that this is something to do with a bad codec negotiation. I'd bet that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip to sip ?
hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns SIP/2.0 404 Not Found any ideas ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel and zaptel versions
On Mon, Feb 19, 2007 at 11:03:17PM -0500, Noah Miller wrote: You can probably solve quality issues without doing a complete reinstall, and you should be able to get reliable results with most all kernel versions and asterisk versions. There are some exceptions, but versions of asterisk and the linux kernel that come with trixbox have generally been tested to work before trixbox gets released. Only CentOS (actually: RHEL) released severarl kernel several kernels since the last -34, and you might wish to check what they fix. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4 svn fails to compile (xbus-core.c)
On Mon, Feb 19, 2007 at 10:50:28PM -0500, Robert La Ferla wrote: This compile error started happening about 2 weeks ago with zaptel. How about looking at my answer to your exactly same question from 5 days ago? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto load of zap drivers
On Tue, Feb 20, 2007 at 11:05:31AM +1100, Klaverstyn, David C wrote: My understanding is that with Asterisk 1.2.x issuing the command of make install-udev allowed the drivers to be loaded upon the server boot. Doing this with version 1.4 does not seem to work. Those udev rules are responsible for the generation of files under /dev/zap/ . Which distribution do you use? Using menuselect I selected zaptel and ztdummy. Should I also be selecting something else for the drivers to load at start up? All you need to run on startup is: modprobe ztdummy Nothing more. Not even a ztcfg. The zaptel init script tries doing that if it senses you have no other zaptel timing source. Do you have ztdummy and zaptel available? modinfo ztdummy modinfo zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Auto load of zap drivers
I am running CentOS 4.4. You say I need modprobe ztdummy on startup. I though the udev option made that happen. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, 20 February 2007 3:11 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Auto load of zap drivers On Tue, Feb 20, 2007 at 11:05:31AM +1100, Klaverstyn, David C wrote: My understanding is that with Asterisk 1.2.x issuing the command of make install-udev allowed the drivers to be loaded upon the server boot. Doing this with version 1.4 does not seem to work. Those udev rules are responsible for the generation of files under /dev/zap/ . Which distribution do you use? Using menuselect I selected zaptel and ztdummy. Should I also be selecting something else for the drivers to load at start up? All you need to run on startup is: modprobe ztdummy Nothing more. Not even a ztcfg. The zaptel init script tries doing that if it senses you have no other zaptel timing source. Do you have ztdummy and zaptel available? modinfo ztdummy modinfo zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for starting point?
On Sun, 2007-02-18 at 14:05 -0500, Gary H. Thompson wrote: Hi, [snip] I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many ‘brands’ of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on books, online tutors, discussions or anything of this nature would be much appreciated. [snip] Welcome Gary. A few suggestions. Get the Asterisk book. You can either order it online (support the cause) or download it for free at http://www.asteriskdocs.org If you don't have prior experience with Linux than you should probably download a Linux distribution that is relatively easy to start with. Two of such distributions that come to mind are Fedora Core 6 and Ubuntu. The advantage of Fedora Core is that there are ready made Asterisk packages available from atrpms.net and laimbock.com/asterisk/ I don't know if that is also the case with Ubuntu. Prepare for a bit of a steep learning curve. If you need help, read the Asterisk book, read pretty much every article on voip-info.org, ask on irc (server freennode.net channel #asterisk) and browse the asterisk mailing list archives. Good luck! Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP response 482 Loop Detected
I solved the problem by making 2 trunks from The Cisco Call Manager using another port 5062 instead of 5060 Also at Asterisk I added a new SIP ... And I modified the extension.conf with these : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) And thus it works ... Thanks ,,, Mohamed Farid ,, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed Farid Sent: Wednesday, February 14, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP response 482 Loop Detected I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... Mohamed Farid ,,, This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for starting point?
On Tue, Feb 20, 2007 at 07:37:02AM +0100, Patrick wrote: On Sun, 2007-02-18 at 14:05 -0500, Gary H. Thompson wrote: Hi, [snip] I understand what a broad scope I am asking about so would appreciate any tips to help me get started. Since there are many ‘brands’ of Linux what is the best one to start with? Which Linux will be better when I get to the point of working with Asterisk? Any tips or ideas on books, online tutors, discussions or anything of this nature would be much appreciated. [snip] Welcome Gary. A few suggestions. Get the Asterisk book. You can either order it online (support the cause) or download it for free at http://www.asteriskdocs.org If you don't have prior experience with Linux than you should probably download a Linux distribution that is relatively easy to start with. Two of such distributions that come to mind are Fedora Core 6 and Ubuntu. The advantage of Fedora Core is that there are ready made Asterisk packages available from atrpms.net and laimbock.com/asterisk/ I don't know if that is also the case with Ubuntu. apt-get install asterisk zaptel-source apt-get install linux-headers-`uname -r` m-a a-i zaptel opensuse also has packages of Asterisk and Zaptel (not to mention Debian, Gentoo and FreeBSD. I only have actual experince with the Debian packages). Prepare for a bit of a steep learning curve. If you need help, read the Asterisk book, read pretty much every article on voip-info.org, ask on irc (server freennode.net channel #asterisk) irc://irc.freenode.net/asterisk should work with decent IRC cleints... and browse the asterisk mailing list archives. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Radius users authentication
Ricardo Carvalho wrote:
Dear all,
I've searched the web about Asterisk with Radius integration for user
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's
Radius client patch, an still open branch of Digium Issue Tracker SIP
peer authentication on an external database (RADIUS - LDAP), etc.
Although, none of these seems to give me the confidence to implement it
in a production environment...
What do you people recommend me? Which Asterisk+Radius solution should
in your opinion be the best choice? Does Asterisk 1.4 already implement
it properly?
Thanks in advance,
Ricardo.
Here is a mock-up of what I used to hook-up to a Radius Server, with Porta's patch. It worked quite
well for us. I have'nt used it in 2 years or so, cant remember much :) . I thin we got it to
work by seeing the debug (set it in /etc/asterisk/logger.conf) and seeing what values were getting
sent and recieved.
;exten = _X.,1,SetVar(RADIUS_Server=x.x.x.x)
exten = _X.,2,SetVar(RADIUS_Secret=secret)
exten = _X.,3,SetVar(NAS_IP_Address=x.x.x.x)
exten = _X.,4,SetVar(CALLERID=${CALLERIDNUM})
exten = _X.,5,SetVar(DNID=${EXTEN})
;
; Set account to authorize by
; It can be a prepaid calling card PIN, ANI, or SIP ID depending on your
application
;
;exten = _X.,6,SetAccount(${CALLERIDNUM})
exten = _X.,6,SetAccount(${CALLERIDNAME})
;
; RADIUS Authorize
; Called as: agi-rad-auth.pl|parametr1=value1parametr2=value2parametr3=value3
; Possible parametrs:
; Routing=XXX will will send h323-ivr-out = 'PortaBilling_Routing:XXX'
attribure (XXX is usually SIP)
; AuthorizeBy=SIP requires SIPGetHeader(SIP_Authorization=Proxy-Authorization) first +
externalauth=yes in sip.conf
; AuthorizeBy=Account requires SetAccount(username) first
; Password=Password optional and may be used together with AuthorizeBy=Account
; IfFailed=DoNotHangup optional, used for custome authentication error
processing i.e. IVR
;
;
exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=${CALLERIDNUM}IfFailed=DoNotHangup
;exten =
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountPassword=AccountIfFailed=DoNotHangup
;exten = _X.,7,agi,agi-rad-auth.pl|AuthorizeBy=AccountIfFailed=DoNotHangup
;
exten = _X.,8,NoOp(${h323-credit-time})
exten = _X.,9, Set(TIMEOUT(absolute)=${h323-credit-time:17})
;exten = _X.,10, AbsoluteTimeout(${h323-credit-time})
exten = _X.,10,Goto(sip-calls,${EXTEN},1)
exten = _X.,11,Hangup
exten = T,1,NoOp(timeout)
--
thanks,
Yusuf
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Re: [asterisk-users] sip to sip ?
create user trunk on each box and dialplan to make call hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns SIP/2.0 404 Not Found any ideas ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
