I am still trying to figure out how to overcome this problem.
I use for International calls a, for USA calls b, ...
Most of the time I get: Forbidden - wrong password on
authentication for INVITE
I would like in that case that the next gateway will be used. How can I
do that?
bye
On Sonntag, 9. April 2006 9:58 Tele Cost Price Reducer wrote:
hi Ronald,
i would use a CallerIDNum authentication, based on the Asterisk
Database to solve it.
then you do not need any verification.
Dangerous. CallerIDs can be easily faked in some countries using VoIP
providers.
Kind
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I have had the exact same problem last week. I have not yet solved it.
So instead I am using ooh323, but would prefer to use oh323. Can anyone
help?
I'm glad that I'm not the only one :))
Hopefully we'll find solution to this
What is the SIP server you specified?
Rudolf
On 4/10/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am still trying to figure out how to overcome this problem.
I use for International calls a, for USA calls b, ...
Most of the time I get: Forbidden - wrong password on
I solved it.
asterisk Ip needs to be binded to HA ip address.
I was using 0.0.0.0 bind ip.
On 9 Apr 2006, at 06:04, Bartosz Wegrzyn - asterisk wrote:
Anyone knows hot to fix that?
Thanks
I used to have my iaxy registered to my old version of asterisk.
I switched to 1.2 ver and now
I can'T compile my
oracle realtime library any more i updatet the svn today and now i tried to
recompile my oracle realtime driver and now it gives me that
errors:
cc -fPIC
-I../asterisk -D_GNU_SOURCE -I/usr/include/oracle/10.1.0.4/client -c
-o res_config_oracle.o
Rudolf Ladyzhenskii wrote:
What is the SIP server you specified?
Rudolf
On 4/10/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am still trying to figure out how to overcome this problem.
I use for International calls a, for USA calls b, ...
Most of the time I get: Forbidden -
Has anyone seen these solid state Drives from gigabyte yet? -
http://www.pcper.com/article.php?aid=224type=expertpid=3
For those who havnt, they are basically a pci card with 4 DDR memory
slots on board, coupled to a SATA interface and with a battery on board
to ensure that you can use
Steve Totaro wrote:
I have no idea what the issues here are, nor do I care but I do have a
question about this statement Since you are selling support for this
script, that qualifies as commercial
use and is expressly prohibited by the micro-license included in the
original script. Is this an
Hi,
How do I read (make sense of) the timestamp in the queue_log? I'm
probably just slow but I don't understand it.
Thanks!
Regards,
Jan
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To UNSUBSCRIBE or update
Darren Ellis wrote:
Hi All,
Could someone send me a code frag on how to get a record from the
asterisk database into a PHP variable via the Manager API?
I can issue calls, etc. from Manager. But I'm not comprehending how to
manipulate database variables.
Google for phpagi, it is a
On 14:30, Mon 10 Apr 06, Ronald Wiplinger wrote:
Which one should I use???
(actually would that be the faster answer)
sip1.voipbuster.com
sip.voipstunt.com
sip.voipdiscount.com
Try to change asterisk useragent in sip.conf.
This has worked for me on several providers.
--
Michiel van
you could use the Action: Command action and pass the cli commands you need, like database show, database put, database deltree, etc...
or the DBput, DBget, DBdel manager actions...Obviously then you have to parse the answers...
Anyway, show manager commands on the cli is your friend... ;-)
An Asterisk box at customer site shows these messages pretty regularly. This
causes one way voice, the called party cannot hear the calling party.
Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for
'0x817b790', 10 retries!
Apr 7 14:47:46 WARNING[18406] channel.c: Avoided
Hi, I have problem with two my asterisk servers (connection by IAX). First is working as master (external dynamic IP with dynamic DNS service). Second is slave behind NAT. The second is registering on first one. Then both servers can make calls to another one.
The problem is that sometimes master
when i load asterisk
i got this error and cant start * with the g729 codec:
Apr 10 10:21:18
VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18 DEBUG[5873]
loader.c: Unexpected signature: 8e 93 22 83 f5 c3 c0 75 ff 8b a9 be 7c 43 74
63Apr 10 10:21:18 WARNING[5873] loader.c: Unexpected key
Has anyone seen these solid state Drives from gigabyte yet? -
http://www.pcper.com/article.php?aid=224type=expertpid=3
Interesting device. Looks like the burst throughput is right on par
with good drives, but you have better sustained throughput and
obviously near zero latency. But what truly
Hello Paul, Thanks for the reply. modprobe gives the following results:
linux:/usr/src/bristuff-0.2.0-RC8q/qozap # modprobe --show-depends qozap
insmod /lib/modules/2.6.13-15-smp/kernel/lib/crc-ccitt.ko
insmod /lib/modules/2.6.13-15-smp/misc/zaptel.ko
insmod
The simplest solution and the one already implemented in linux, tmpfs.
It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel
do the work it was designed to do. And you would not be limited to PCI
bus speeds. The DDR2800 is about 12GB/sec. Some would say overheads,
etc, etc.
On Sun, 2006-04-09 at 13:20 +0200, Paul Hewlett wrote:
On Saturday 08 April 2006 20:18, Colin MacMillan wrote:
Hello,
6) From here I enter the qozap directory. cd qozap
7) now I get the following error -
Hi,
I've just come upon an interesting question regarding the use of Asterisk as an
Application Server connected behind a conventional ISDN PBX: The user wants
to forward all incoming calls through the PBX to Asterisk over S0-Lines, have
Asterisk do some processing (which includes looking up
On 4/10/06, Dave Cotton [EMAIL PROTECTED] wrote:
This has all the hallmarks of the kernel source and actual runningkernel not being the same.--Dave Cotton [EMAIL PROTECTED]
Hello
Dave, This is a fresh install of Suse Linux 10 from the CD so I
can't see how the source and running kernel differ. One
Hi,
when I try to use meetme I always hear this error message
this is not a valid conference number, please try again,
but my configuration seems to be correct... Here it is:
-- extensions.conf --
exten = 6000,1,MeetMe(1234,ciMp) ; entra nella meetme room 1234
-- meetme.conf --
[rooms]
conf =
Hi,
I wrote a script for making a gsm or wav file out of a text, i tried it
first with text2speech from the festival package, and sox for converting
it to a gsm file, that worked really well... and I can play it back with
the Playback cmd.
But only in English, so I tried Mbrola with a German
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
sip:[EMAIL PROTECTED] (french)
Thanks for help
___
Nouveau : téléphonez
On Monday 10 April 2006 11:59, [EMAIL PROTECTED] wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
sip:[EMAIL PROTECTED] (french)
Hi,
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
No Sound or voice!
Thanks Thomas,
I could not hear you too !
may be the firewall
Harry
--- Thomas Winter [EMAIL PROTECTED] a écrit :
On Monday 10 April 2006 11:59, [EMAIL PROTECTED]
wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music
Somebody can say me what i can do that the g729 is
working?
Von: René Enskat [Teamware GmbH]
[mailto:[EMAIL PROTECTED] Gesendet: Montag, 10. April 2006
10:21An: 'asterisk-users@lists.digium.com'Betreff: G729a
error
when i load asterisk
i got this error and cant start * with the g729 codec:
I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
configured using the provisioning feature so the configuration is all
identical.
The problem I am having is that they randomly seem to stop registering
with asterisk. When they stop registering they can still make calls but
Hello,,
I have some of asteriskto CCM4 SIP trunk oneway audio problems.
I have setup a asterisk server. It's work great and have no
any problemsconnect to local ITSP(using SIP protocol). But we need to
build a sip trunk to another CCM4 server.
The network typology like this.
SIP Phone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I'm in charge of two Asterisk servers.
One of these servers is running an [EMAIL PROTECTED] version under CentOS,
the other one is running Debian Sarge. The Debian server has been
installed by me using this howto:
Yes. Me.
I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch.
I would definitely recommend it to you.
Mark
-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED]
Sent: Monday, 10
Peter J Dean wrote:
I have an issue with trying to ensure that when dialling an extension
that it continues to ring up to the timeout value. But what I am finding
is that the timeout is all over the place. Sometimes half the timeout
value and other times within a few seconds of the timeout
Hi,
I have the same problem on TE110P and Taiwan telco PRI line. I think to fine
tune the rigntone frequencies not resolve this problem.
For example.
When I make a call to mobile. I can hear one ringtone like geneate by
asterisk or device. And another ringtone like from telco. You known, some
Very well said and I couldn't agree more!!!
Rich
The VoIP Connection wrote:
Not true. There are hundreds of thousands of Grandstream adapters in
use around the world. Grandstream support is not perfect, but it is as
good or better better than most vendors, including Linksys/Sipura. The
I found progressinband=no in sip.conf fixed my problem when I had this.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: 10 April 2006 12:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Mon, 10 Apr 2006, Lee Archer wrote:
I found progressinband=no in sip.conf fixed my problem when I had this.
Regards
Lee
I can confirm this solution has worked for me. It is a know bug (no.
6690), see http://bugs.digium.com/view.php?id=6690
I managed to get it to work and make a test call from Asterisk to Skype.
Pity it's not implemented on Linux and needs Skype to be running on the
PC also.
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent: 05 April 2006 16:18
To:
Hi
Thanks a lot. It's work for my ArtDio IPF-3000 phone. I have make a lot fine
tune on the zapata.conf file. Doesn't have any help.
Just add progressinband=no in the sip.conf. Done!
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
So the bug still exists in the 1.0.2 branch?
Thanks
On Mon, 2006-04-10 at 12:14, Mark Edwards wrote:
Yes. Me.
I don't have a fix unfortunately - like you I seek one, however I have had a
better experience by far though with the new 102x firmware branch.
I would definitely recommend it
Could you try again please?
--- Thomas Winter [EMAIL PROTECTED] a écrit :
On Monday 10 April 2006 11:59, [EMAIL PROTECTED]
wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
sip:[EMAIL PROTECTED] (french)
Hi,
The CCM4 behide the PIX firewall?Have you open the
ports for SIP trunk on CCM4 side? (TCP/UDP 5060, UDP
16348-32768)
Kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio
Serrano LuqueSent: Wednesday, March 01, 2006 4:18 PMTo:
On Mon, 10 Apr 2006 12:41:21 +0100, Whisker, Peter wrote:
I managed to get it to work and make a test call from Asterisk to Skype.
Pity it's not implemented on Linux and needs Skype to be running on the
PC also.
Peter
This is the same with the PSGW product. It's the only way to
I've also had horrible experiences with the Asterisk plugin. The second I enable it, no one can log into their IM client anymore.KyleOn 4/9/06, Kerry Garrison
[EMAIL PROTECTED] wrote:
I tried the latest version of Jive over the weekend and I have to say it isa giant pile of crap. I did this on
This is working now thanks to Roelof Dijkstra's instructions below. Thank you Roelof!
Hi
Colin,
What i
did , was that i added qozap, to the Makefile of zaptel.
I just
copied the files of the qozap directory there, and added the line in the
Makefile.
After
that , i could
Hi, My * refuses SIP registrations when internet is down. All is returing at the moment when outside connection is up. What is wrong?-- Best regards,Michael Strelnikov
___
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Asterisk-Users mailing
On Monday 10 April 2006 13:49, [EMAIL PROTECTED] wrote:
Could you try again please?
--- Thomas Winter [EMAIL PROTECTED] a écrit :
On Monday 10 April 2006 11:59, [EMAIL PROTECTED]
wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
Tomislav,I have asterisk-1.2.5 and asterisk-oh323-0.7.3goksieOn 4/10/06, Goke Aruna [EMAIL PROTECTED]
wrote:I have asterisk-1.2.5 and asterisk-oh323-0.7.3
On 4/10/06, Tomislav Parčina
[EMAIL PROTECTED] wrote:
Hi!I have one short question for you. Can you tell me what version of Asterisk do you
In article [EMAIL PROTECTED],
Giuseppe [EMAIL PROTECTED] wrote:
Hi,
when I try to use meetme I always hear this error message
this is not a valid conference number, please try again,
but my configuration seems to be correct... Here it is:
-- extensions.conf --
exten =
On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote:
Hi,
My * refuses SIP registrations when internet is down. All is returing at
the moment when outside connection is up. What is wrong?
Try to set srvlookup=no in your sip.conf
Or put all the phone ip's in the servers /etc/hosts
This is
Just wondering everyones experience with Six Tel (http://www.iax.cc/show.php?go=local)?
They seem to have some really decent prices but I have heard some buyer
beware comments elsewhere.
Thanks
Curt
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I switched PRI vendors recently, and one of my questions was do you provide
caller ID name in addition to number?
ATT Local did not, But XO communications said they did.
Before I call to complain, is there an setting to turn this on in asterisk?
I want to make sure that I have my side covered
I have a number of toll free DIDs with
them. Never had a problem. Control panel is excellent and they even give you
specific configs. I dont run much traffic through them but when I do,
no problems. As always, let the buyer beware.
Thanks,
Steve Totaro
From: Curt
How is the 9133i configured, through the .cfg file, the WebUI, or the
Phone's own interface? The PhoneUI WebUI take precedence over the
.cfg file.
You can look at the WebUI and see what the current settings are, and
clear them out if you'd rather use the .cfg file settings.
-Original
That's a standard unix timestamp: # of seconds from Jan 1, 1970.
l.
In data Mon, 10 Apr 2006 08:55:49 +0200, [EMAIL PROTECTED] ha scritto:
Hi,
How do I read (make sense of) the timestamp in the queue_log? I'm
probably just slow but I don't understand it.
Thanks!
Regards,
Jan
--
Assum
Kyle Sexton wrote:
I've also had horrible experiences with the Asterisk plugin. The second
I enable it, no one can log into their IM client anymore.
did you report that on their forum?
I installed it some time ago and it worked quite well besides some
issues with staying on the phone when
Thomas Broda wrote:
Which component do I have to install in order to get the Maintenance
setup?
The Maintenance tab is part of [EMAIL PROTECTED] and not AMP/freePBX. You'll
only see it on an [EMAIL PROTECTED] installation.
cYa,
Avi
--
National Manager - Special Projects
Melbourne /
I will implement a SIP application and I'm considering using Asterisk for mixing the media
streams (audio). Does anybody know if Asterisk
supports or contains a RTP mixer? If so, how to use it?
Just to be a little more clearer: I will send to Asterisk more
than one RTP stream and they must be
Hi,I've been watching my * Console and seems to be one call not well terminated or something:For 5 minutes at least my console is reporting this: ectory|general|ext-local|be: -- Playing 'letters/c' (language 'en')
directory|general|ext-local|be: -- Playing 'letters/o' (language 'en')
René Enskat [Teamware GmbH] wrote:
Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18
DEBUG[5873] loader.c: Unexpected signature: 8e 93 22 83 f5 c3 c0 75 ff
8b a9 be 7c 43 74 63
Apr 10 10:21:18 WARNING[5873] loader.c: Unexpected key returned by
module
I will implement a SIP application and I'm considering using Asterisk for mixing the media
streams (audio). Does anybody know if Asterisk
supports or contains a RTP mixer? If so, how to use it?
Just to be a little more clearer: I will send to Asterisk more
than one RTP stream and they must be
Thomas,The maintenance tab is coded by Andew Gillis (owner of the [EMAIL PROTECTED] project) specifically for AAH. It's typically difficult to replicate on non-AAH systems due to missing software packages. If you want to try, you need to take a look at the PHP page that the maintenance tab uses,
Hi,
I am using Asterisk fora call center on a
Dual Xeon machine..
I currently have
109 active channels
53 active calls
Every body is complaining about quality and cpu is
around 80% idle.
Is there any tuning I can do???
Besides that, Asterisk normally goes down once
or twice per day...
Hi found that it could happen just using Xlite and after dialing *411 , then change your Xlite to line2 without hanging up channel 1 My solution has been on CLI a soft hangup for the SIP channel that made this call.
I found the channel with show channels.On 4/10/06, Marco Mouta [EMAIL
I have this to access directory of Asterisk:exten = *411,1,Answerexten = *411,2,Wait(1)exten = *411,3,AGI(directory,general,ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})exten = *411,4,Playback(vm-goodbye)
exten = *411,5,HangupWhich is the safest way to establish an Absolut Timeout for this?exten =
Steve Totaro wrote:
I have a number of toll free DIDs with them. Never had a problem.
Control panel is excellent and they even give you specific configs. I
don’t run much traffic through them but when I do, no problems. As
always, let the buyer beware.
Thanks,
Steve Totaro
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
On Thu, 06 Apr 2006 13:17:29 +0200, Tomislav ParÄŤina [EMAIL PROTECTED]
wrote:
How did you modify it? and will the ATXFR be perceived as a discharge from
the queue system as a blind transfer using #?
features.conf
blindxfer = #1
Marco Mouta wrote:
I have this to access directory of Asterisk:
exten = *411,1,Answer
exten = *411,2,Wait(1)
exten =
*411,3,AGI(directory,general,ext-local,${DIRECTORY:0:1}${DIRECTORY_OPTS})
exten = *411,4,Playback(vm-goodbye)
exten = *411,5,Hangup
The Asterisk directory is an
Hi all, a noob here, I am trying to get outbound calls through asterisk
working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in an effort to make
I have had DIDs and termination with sixtel for over a year. No problems
here.
Cheers,
Zac
--
Zac Amsler, Network Operations
NetIQ Systems, LLC www.netiqsys.net
Wholesale VOIP Termination.
Jay Milk wrote:
Probably one of the best examples of your mileage may vary and
buyer beware.
When
Hi i tried this setup what should my zapata.conf setup be?
On 4/8/06, Infobox Peru
[EMAIL PROTECTED] wrote:
Your zaptel is wrong...
it must be:
zaptel.conf:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31On 4/7/06, JP Carballo
[EMAIL PROTECTED] wrote:
Mark Quitoriano wrote: Hi Guys, Im
Hi list,
I know * generates it's outgoing RTP stream based on the incomming RTP stream,
i'm having some audio problems after i recieve an rtp reinvite from my
carrier.
Situation:
Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN
Asterisk A: reinvite = no
Asterisk B: reinvite = no
If i
Erik wrote:
IMHO Asterisk B should change it SSRC to tell Asterisk A the RTP source has
changed (and fix this timestamp gap)?
That's an interesting question; since Asterisk is not actually a proxy,
in point of fact the SSRC has _not_ changed, since Asterisk B is still
the source of the RTP
There is nothing you really need 'to do' if your PRI is working already,
If you are able to receive and make calls your D-Channel is functioning
properly. In the case of CallerID, some telcos provide this extra
function via the FACILITY messages instead of the SETUP messages, If
that is the case,
I have noticed other parts of the U.S. and the world posting about user
groups, and wondered if we have enough Asterisk users and interest in
the Carolinas to start a group. I am willing to help with the
organizational effort if there is interest and maybe one or two other
folks who would be
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's in my config or not (if
From what you say it sounds that the problem is not with asteisk, but
the way it's configured. Asterisk should *never* go down that often.
Asterisk as a normal PBX should run without a restart for as long as
there is power to the box, in the case of a call center if I would
hear of a restart once
Dov,Do you have any quality of service capability on your data network for prioritising voice? With that volume of calls it could be a network load issue if your cpu seems to be fine.What kind of interface are you using to the PSTN?
On 4/10/06, Dov Bigio [EMAIL PROTECTED] wrote:
Hi,
I am
Can someone explain me this message:
chan_iax2.c: Ooh, voice format changed to ...
Where can I find a list of numeric codes used to identify voice format?
Then, sometime I get an infinite loop of messages like these:
DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1
WARNING[15015]
On 4/10/06 12:25 PM, Mimmus [EMAIL PROTECTED] wrote:
Can someone explain me this message:
chan_iax2.c: Ooh, voice format changed to ...
Where can I find a list of numeric codes used to identify voice format?
Then, sometime I get an infinite loop of messages like these:
DEBUG[15015]
Besides that, Asterisk normally goes down once or twice per day...
sounds like a standard mutex deadlocks ???
you mean the cli is unresponsive, no new calls are accepted
OR
do you mean a real core dump thats much easier to resolve
just get a bt post to the end less bugs at bugs.digium.com
you
How do I setup faxing in asterisk
Corne
Labuschagne
SAIL Communications
Tel: +27 16 422 8195
Fax: +27 16 422 8196
Cell: +27 82 851 6893
Mail: [EMAIL PROTECTED]
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Hello everyone - first time poster, long time lurker. (sounds like a
radio morning program, I know).
I'm attempting to get my InterTel Axxess (w/v9.0 software) to play nice
with my Asterisk implementation. Asterisk 1.2.6 is running on a Fedora
Core 4 x86 box. I've tried getting the Axxess to
Corne Labuschagne wrote:
How do I setup faxing in asterisk
http://tinyurl.com/qddpf
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
we're using aheeva on a call center on Dominican Republic, we're using
aheeva since 2 month,
Wai Wu wrote:
Isn't aheeva a commercial product? Whoever wants to find out how it is should
ask aheeva for referrals, and I recommand him personally pay visits to their
customers on their expenses
I'm
wondering if the page application is broken in 1.2.5
The
following:
exten
= 2001,1,Page(SIP/3254105)
does
strange stuff. The caller's phone immediately drops into the call, while the
callee's phone is still ringing. I'd think it was a SIP messaging issue, except
that the Dial()
Douglas Garstang wrote:
exten = 2001,1,Page(SIP/3254105)
does strange stuff. The caller's phone immediately drops into the call, while
the callee's phone is still ringing. I'd think it was a SIP messaging issue,
except that the Dial() command is working fine, which makes me wonder if
OK I am going to do it again.
Global Crossing is now sending ANI but it is not in the format I expected. Any
one know of a way to get this data into two seperate variables? The first
number is ANI and the second is DNIS so it is *tendigits*tendigits* on one
line like below.
Called
please read archives! and voip-info.org
this question appeared so many times over here and was replyed so many times
too!
- Original Message -
From: Corne Labuschagne
To: asterisk-users@lists.digium.com
Sent: Monday, April 10, 2006 12:37 PM
Subject: [Asterisk-Users] faxing setup
Mike Raley wrote:
Hi all, a noob here, I am trying to get outbound calls through
asterisk working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in
I am still looking for a solution and I am sure that I am not the only
one having that problem:
If provider A fails for any reason, the next provider should be taken.
There are many reasons, why a provider fails, like:
password wrong (cli reports so, but actually it is the gateway's problem)
I am trying to figure out where the problem for one direction dialing
results in audio or no audio!
Please help me to compile a list of possible setting errors.
My server has two Ethernet connections, one to the Internet, one to the
local LAN (NAT).
The remote user can call phones in my
Ronald Wiplinger wrote:
I am still looking for a solution and I am sure that I am not the only
one having that problem:
If provider A fails for any reason, the next provider should be taken.
There are many reasons, why a provider fails, like:
password wrong (cli reports so, but actually
Please look at:
http://www.sineapps.com/news.php?rssid=1130
SNIP...
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Title: Re: [Asterisk-Users] ANI on a PRI
Is this on a PRI???
On a RBS T1 this is called "Feature Group D (Adtran
Style)"
If they are sending you this via a PRI they may have
their Trunk Group setup wrong, as I have never seen this type or data element
structure on a PRI.
.
But in a pinch
Our user places a call, the gateway responds with no sound at all, or
hangs up, or gives busy tone.
How can we get to the next provider?
I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten =
Thanks Ronald!
Works like a charm. What my brain sees and what my hands type are not always
the same thing! doh!
MR
maybe just a typo, br_OA_dvoice
I gone away from broadvoice, since they admitted to have troubles and I
had still to pay for NO phone call !!! (multiple lines)
bye
Yes its a T3 split into seven trunk groups with one D channel and NFAS on each.
Can you explain the cut function or point me somewhere please?
Thanks,
Steve
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Mon 4/10/2006 12:51 PM
Title: Re: [Asterisk-Users] ANI on a PRI
from the CLI
show function CUT
or if running on pre 1.2
show application cut
SNIP...
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Brent Torrenga wrote:
Our user places a call, the gateway responds with no sound at all, or
hangs up, or gives busy tone.
How can we get to the next provider?
I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
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