Re: [asterisk-users] Ultramonkey LVS + asterisk
Hey Ron, Did you get your ultramonkey setup working correctly? I'm about to roll ultramonkey here, any tips? Regards, Igor H. Nhadie wrote: hi, has anyone implemented ultramonkey with asterisk? do i really need to setup fwmark as discussed in the url below? thanks! http://www.gossamer-threads.com/lists/lvs/users/20871 regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra phones and dns srv records
Hi guys, Does the Aastra line of phones work with dns srv records? I'm trying to get my 8133i to do this and in the settings it asks for ip addresses of registration and proxy servers. Does this mean that it will not just let me put the domain name in like other devices I have and then do fail over to other servers when needed? If these phones do not what phones do? Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add Callee's name into Dial command ?
2008/10/3 satish patel [EMAIL PROTECTED]
2008/10/3 Joe Pukepail [EMAIL PROTECTED]
I think this is what you want: http://bugs.digium.com/view.php?id=8824
Thanks : this one very interesting.
Bottom line is it doesn't work at the moment right ?
http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include this kind
of line :
To: sip [EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like this
instead ?
To: John Doe sip [EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED];user=phone
Regards
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why you people need this thing in dial command which can possible
with sip.conf callerid options
Unfortunately, callerid option in sip.conf is not used to callee's name in
caller's phone screen :
if Alice calls Bob, Alice's phone will display Bob's number but not Bob (ie
callee's name)
If you SIP messages that comes back from Asterisk to Alice's phone, you
won't find the name Bob anywhere, so obviously, as Alice phone will use
those messages to update its own screen, you won't see any sign of callee's
name anywhere.
P-Asserted-Identity is a rather new field which is dedicated to such names
and is supported by several phones.
At the moment, Asterisk won't add this field in any reply to Alice's INVITE.
Cheers
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Re: [asterisk-users] 2 stage dialing and 484 address incomplete [SOLVED]
Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier [EMAIL PROTECTED] Hi, If my memory serves me right, there was thread (in dev mailing list ?) explaining how we could implement 2 stages dialing with SIP endpoints: user dials 1234 then asterisk replies 484 Address Incomplete, then user dials 5678 then asterisk begins to treat extension 12345678 as if it had been dialed as a whole. With compliant hardphones, you could get you phone to display a short text invite between the series of digits. This improves user experience, when consulting Voicemail, or asking features that need a parameter to be set. Is my memory correct ? Has anyone a clue ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uninstalling zaptel
Hello. Go to Zaptel dir and type make uninstall make uninstall all make remove Before removing Zaptel, be sure Zaptel is stopped. /etc/init.d/zaptel stop There are some files which not removed by make. If necessary, you can delete these files manually. But if Zaptel is not loaded, it's not necessary. On Fri, Oct 3, 2008 at 4:39 AM, Jerry Geis [EMAIL PROTECTED] wrote: What is the correct way to uninstall zaptel in the zaptel directory I can do make uninstall-modules which does just that but what about all the other files??? /etc/udev/rules/XX /etc/init.d/XX /sbin/ztXX and others doing a make uninstall gives an error. Is there anything that removes all those other files. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
Hello.
Have you ever tried updating your GCC version?
Thanks.
On Thu, Oct 2, 2008 at 8:30 PM, Satish Patel [EMAIL PROTECTED] wrote:
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
As I wrote:
Could you please try a newer version of zaptel 1.4? There have been
many
changes in the build system of zaptel 1.4 since 1.4.1 .
But in your reply:
clfs:/mnt/clfs/sources/zaptel-1.4.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
I wanted to show you what option i used now i have download
zaptel-1.4.12.1
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
configure: WARNING: If you wanted to set the --build type, don't use
--host.
If a cross compiler is detected then cross compile mode will be
used.
I don't know much about cross-compiling, but this warning scares me.
I have a feeling you're doing something wrong.
Anyway, if you want to avoid the whole menuselect mess, take a look at
http://bugs.digium.com/13132
Remove the subdirectory menuselect and put the makefile and script from
that bug report there instead. Run:
make -C menuselect dummies
Then it should behave just like the original. At least theoretically.
You may need to instruct it to take data from other XML files. See the
calls to the function parse_menuselect_xml_file() in the end.
Let me know if it worked ;-)
I'll see if someone else will pick it up on-list as both cross-compiling
and menuselect are not my preffered code.
for experiment i have download 1.2.27 current version of zaptel
./configure --host=${CLFS_TARGET} --prefix=/usr
make ARCH=arm CROSS_COMPILE=${CLFS_TARGET}-
make ARCH=arm CROSS_COMPILE=${CLFS_TARGET}- DESTDIR=${CLFS} install
it has installed module
clfs:/mnt/clfs/sources/zaptel-1.2.27$ ls -l ../../lib/modules/
2.6.22.6/misc/
total 476
-rw-r--r-- 1 clfs clfs 67566 Sep 1 18:46 pciradio.ko
-rw-r--r-- 1 clfs clfs 92753 Sep 1 18:46 tor2.ko
-rw-r--r-- 1 clfs clfs 19267 Sep 1 18:46 torisa.ko
-rw-r--r-- 1 clfs clfs 15542 Sep 1 18:46 wcfxo.ko
-rw-r--r-- 1 clfs clfs 18524 Sep 1 18:46 wct1xxp.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wct4xxp
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wctc4xxp
-rw-r--r-- 1 clfs clfs 46475 Sep 1 18:46 wctdm.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wctdm24xxp
-rw-r--r-- 1 clfs clfs 40601 Sep 1 18:46 wcte11xp.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wcte12xp
-rw-r--r-- 1 clfs clfs 18531 Sep 1 18:46 wcusb.ko
-rw-r--r-- 1 clfs clfs 71372 Sep 1 18:46 zaptel.ko
-rw-r--r-- 1 clfs clfs 8250 Sep 1 18:46 ztd-eth.ko
-rw-r--r-- 1 clfs clfs 4883 Sep 1 18:46 ztd-loc.ko
-rw-r--r-- 1 clfs clfs 3204 Sep 1 18:46 ztdummy.ko
-rw-r--r-- 1 clfs clfs 13059 Sep 1 18:46 ztdynamic.ko
-rw-r--r-- 1 clfs clfs 10780 Sep 1 18:46 zttranscode.ko
but when i have build-root and run is root-image on ARM hardware and
try to install module i got error
root#insmod zaptel
insmod: cannot insert '/lib/modules/2.6.22.6/misc/zaptel.ko' : Invalid
module formate (-1): Exec format error
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[asterisk-users] Mimic SIP Events framework in Asterisk without coding ...
Hi, You can see here and there, several new SIP RFCs relying on SIP Events Framework. For example, RFC3680 with which a registration server would notify endpoints with relevant events. In Asterisk 1.6.1, a new SIPnotify AMI command implements a mechanism to send arbitrary NOTIFY commands. Is there any sister SUBSCRIBE mechanism that allow an application relying on AMI to receive SUSCRIBE messages matching some criteria ? With both NOTIFY and SUBSCRIBE tools in hands, one could extend Asterisk to support SIP RFC without having to code into Asterisk source. Your thoughts ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: sip clients for smart phones?
You realy have issues .. instead of wasting my time and the group's time and your own time with such emails.. just ignor my emails from now on.. i've been in this list for years now.. you are the first one who spoke of this .. and you want me to change my email address?? how smart is that? live with the HOTMAIL rules man! regards AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 4 Oct 2008 02:56:34 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OT: Re: sip clients for smart phones? Tariq .. schrieb: i'm using Hotmail webmail.. so what is wrong with it? See for yourself: http://lists.digium.com/pipermail/asterisk-users/2008-October/219531.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html Ironically you would think that *my* email client is broken while actually yours messes up the text/plain part. And in addition to that - You don't skip the irrelevant parts (e.g. my signature or the list footer) - You top-post - You violate email netiquette by not using a proper signature separator - You send a footer telling me about Windows Live which is totally unrelated. Even for free email accounts that's not acceptable any longer since there are free accounts without advertising. I could live with 1 or maybe 2 of these issues but 5 is a bit much. You didn't even notice these problems, so, ok, sorry for being rude. But for people who are used to email in ages it feels like a punch in the face. It's a real culture clash. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ See how Windows connects the people, information, and fun that are part of your life. http://clk.atdmt.com/MRT/go/msnnkwxp1020093175mrt/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mimic SIP Events framework in Asterisk without coding ...
On Sat, Oct 04, 2008 at 02:02:48PM +0200, Olivier wrote: Hi, You can see here and there, several new SIP RFCs relying on SIP Events Framework. For example, RFC3680 with which a registration server would notify endpoints with relevant events. In Asterisk 1.6.1, a new SIPnotify AMI command implements a mechanism to send arbitrary NOTIFY commands. Is there any sister SUBSCRIBE mechanism that allow an application relying on AMI to receive SUSCRIBE messages matching some criteria ? With both NOTIFY and SUBSCRIBE tools in hands, one could extend Asterisk to support SIP RFC without having to code into Asterisk source. Use sipsak (apt-get install sipsak) ? This program has to be good for something. AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net I tried. I get: Comming soon, registration for AstriCon 2009. :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip clients for smart phones?
Yup. Did that in the same setting flow, yet it didn't show up when TF3D was off. Oh well. Someone said use Fring and I think so far it's worked over EVDO. Nice! Thanks.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 4:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? Ah, I don't use the touchflow crap :) On mine on the today screen (you'll have to go to settings, today, items) Set internet telephony to on and you should see it on the home screen. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I don't even see it anywhere on TouchFlo3D. I don't see where to even use it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? I use the TytnII with Win Mob 6.1, customized ROM and it's working for me - through the back speaker though. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 03, 2008 2:57 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This doesn't work in WinMo6.1 for some reason. Especially on touchscreen phones. Touch Diamond for instance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: October 3, 2008 11:52 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip clients for smart phones? This may help: http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo ws-mobile-6x-for-free-voip-calls-using-asterisk/ Note, that most sip clients for WINMOB suck and send the voice out the back speaker instead of the front speaker. I've found one other client (can't remember the name now) that works through the back speaker, but it's payware, and nowhere near as lightweight as this method. For now I deal with the back speaker. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, October 03, 2008 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip clients for smart phones? On Fri, 3 Oct 2008, Babcock, Michael Alex wrote: hi; curious does anyone know of a smart phone client that could connect to asterisk? All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I use an E90 over Wi-Fi. Phone gets hot and battery life is about an 1.5 hours of talk time, but... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] IAX denial of service
How can I prevent a remote DoS as described on the following site? : http://www.voip0day.com/news/remote-denial-of-service-exploit-effects-the-asterisk-pbx/ Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.manta.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the full configurations
that are available are a bit complex. Any tips or recommendations to
get up and running would be great.
sip.conf
Code:
[general]
register = rsreese:[EMAIL PROTECTED]:5060
context=default ; Default context for incoming calls
realm=ns1.neocipher.net ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
domain=neocipher.net; Set default domain for this host
[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=test81
mailbox=101
callerid=Stephen 101
host=dynamic
dtmfmode=rfc2833
canreinvite=no
reinvite=no
disallow=all
allow=gsm
[102]
type=friend ; allows incoming and outgoing calls
username=102
secret=test81
mailbox=102
callerid=(Bob 101)
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
allowguest=yes
insecure=very
promiscredir=yes
musicclass=default ; Sets the default music on hold class
for all SIP calls
[authentication]
[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=pass
allow=all
insecure=very
canreinvite=no
[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=pass
allow=all
canreinvite=no
extensions.conf
Code:
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail(102)
exten = 102,1,Dial(SIP/102,20)
exten = 102,2,Voicemail(102)
exten=*98,1,VoiceMailMain([EMAIL PROTECTED]) ;This
automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])
[inbound]
exten = 9045622082,1,Answer
voicemail.conf
Code:
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes ; Context to Send voicemail from [option 5
from the advanced menu]
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
[default]
101 = 123,Stephen Rese,[EMAIL PROTECTED]
102 = 123,Bob Dole,[EMAIL PROTECTED]
1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
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Re: [asterisk-users] Improving the voice Quality,
Steve, Steve Totaro wrote: BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp built right in, as well as other great apps, utilities, and security auditing. I suggest everyone have a copy in their arsenal, and it is free of course. What does it do? I mean, for someone like you, practically speaking? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail quota
Hi, I am using asterisk-1.4.11. Voicemail quotas only apply to the new messages in the INBOX. Browsing quickly through the 1.6 app_voicemail it seems that 1.6 does implement voicemail quota for both INBOX and Old messages. Is that correct? If so, is there an existing patch available that anyone would know of? Thanks, Sebastien. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? --- On Fri, 26/9/08, Darrick Hartman [EMAIL PROTECTED] wrote: From: Darrick Hartman [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:52 AM ...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Load Balancing
Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
OpenSIPS/Kamailio have modules designed specifically for that kind of functionality now without a need for an outside monitoring process or SRV reliance. Darren Sessions wrote: One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balancing
I know. :) I've already mentioned some of the OpenSIPS options to him on the OpenSIPS users list (LCR module specifically). Just brain dumping everything that came to mind. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 10:31 PM, Alex Balashov wrote: OpenSIPS/Kamailio have modules designed specifically for that kind of functionality now without a need for an outside monitoring process or SRV reliance. Darren Sessions wrote: One other thing you could try would be to use OpenSIPS and use a standard config that routes to a hostname (with a creative failure route setup). You'd then setup the hostname in DNS as multiple SRV records reflecting your pool of Asterisk servers (set your TTL very low for these records). You could have something like sipsak send test messages every 30 seconds or so to each of the Asterisk servers. If one quits responding, then the monitoring app updates your DNS servers removing the effected Asterisk server from the DNS pool and effectively from the usable gateway pool. I actually wrote one of these ages ago that worked fairly well with a10 calls per second SER server. How many calls per second are you looking to process? - D _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 4, 2008, at 9:59 PM, John D wrote: Hi all, I've googled around for concrete solutions on load balancing Asterisk, and it appears there are several ways to skin this cat -- but not one solution which is all appealing. I have the following requirements, which aren't anything extraordinary: * I need to handle roughly 300 simultaneous phone calls to start * Eventually scale to 1000 simultaneous phone calls * I want to be able to pull out an entire server from the cluster without affecting my application * I'm doing all my trunking over SIP So far I've seen folks mention the use of DUNDi and OpenSER(Now OpenSIPS), but unfortunately the documentation out there is rather sparse and lacks detail for someone who isn't extremely keen with the intricate details of Asterisk or OpenSIPS. Would anyone be able to suggest a good starting point in as far as reading documentation and testing out some solutions? I'd also be up for hiring a consultant to help me get started -- but I believe the proper forum for that is asterisk-biz. (Which I've already posted to). Thank you for your insight on load balancing Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
