At 08:47 PM 7/23/2007, you wrote:
For both lines I can definitely see the callerid being reported correctly
in /var/log/asterisk/cdr-custom/Master.csv. It's just not getting passed
through to the sip clients for the incoming line with distinctive ring.
For those lines it shows up as asterisk.
FERNANDO VILLARROEL schrieb:
--- Knud Müller [EMAIL PROTECTED] wrote:
FERNANDO VILLARROEL schrieb:
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider
a
real phone number (ISDN phone or
Am Montag, den 23.07.2007, 06:44 -0700 schrieb satish patel:
Dear all
I have configure asterisk with 100 SIP PHONE ( SNOM )
but now thing is that my boss need phonebook feature find extention
number by Pbook so i have read about it there is a feature in asterisk
but it is
Hi,
I'm trying to set a rule to dial out through multiple
Zap groups so that, say, g0 is the cheaper POTS lines
group
and must be used first. However, if g0 is busy or
disconnected then try dialing out g1.
My g0 group is made up of 4 analog lines connected to
a 4-FXO card. I disconnected the
I have a Diva Server BRI installed in a Debian system
running Asterisk 1.2.22. When a call comes in the
Background application plays a couple of times.
Halfway through the second time, the call is
disconnected (total time connected about 25 seconds).
I've posted various configs below. Any advice
Hello,
again, as discussed in the other Thread, these are SI-120. The SI-120 is NOT
a snom phone, it is a cheap phone sold by our subsidary in India.
Neither the Software nor the Hardware was German engineered by snom
technology AG. Thus there is no Minibrowser in this phone.
To prevent
On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote:
On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
Hi,
Frozen or crashed? Do you see the console of the system?
serial console is dead.
kernel is 2.6.18-4 debian Etch.
bristuff is latest zaptel-1.2.19 and asterisk-1.2.22
I
Noah Miller wrote:
I'm looking for 24 or 48 port IEEE802.3af POE injector.
Any recommendation?
Yes. For the price of one of those multi-port injectors, you can come
close to the price of a new Netgear or 3Com PoE switch. The injectors
typically add power to the unused pairs (mode B PoE).
On Tue, Jul 24, 2007 at 12:05:47PM +0200, Thomas Winter wrote:
On Thursday 19 July 2007 04:27, Tzafrir Cohen wrote:
On Wed, Jul 18, 2007 at 12:44:29AM +0200, Thomas Winter wrote:
Hi,
Frozen or crashed? Do you see the console of the system?
serial console is dead.
kernel is 2.6.18-4
IEEE802.3af uses same 4 wire as data.
thats what Polycom uses.
the way i'm seeing it we are better off with poe switch(looking at the
price).
802.3af has two different modes:
Mode A: uses the same 4 wires as 10/100 ethernet, typically done by
PoE endpoints like switches
Mode B: uses the
Hi Mike -
It seems like since we got FIOS
installed (including switching to fios phone lines which are supposed to be
the same on our end) i am having massive problems with asterisk not hanging
up dead calls for days, even weeks if i dont catch it. It slowly builds up
randomly not ending a
I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm
following the instructions posted here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
I have MySQL installed and it works fine - starts on stratup, I can
create DBs, tables and so on and I can connect through php. rpm -qa
Hi Vieri -
I'm trying to set a rule to dial out through multiple
Zap groups so that, say, g0 is the cheaper POTS lines
group
and must be used first. However, if g0 is busy or
disconnected then try dialing out g1.
My g0 group is made up of 4 analog lines connected to
a 4-FXO card. I
1 there is a correct file in /var/spool/asterisk/outgoing
2 i run asterisk -r to monitor it , it gives out the following error
-- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)
Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
Hello
We use the 2BRI version of Astribank in production, and it has been working non
stop for about amonth now, without any problems.
It was a bit difficult to setup, but other than that, it was great.
Great concept with using the USB2 port for channel banks.
Regards
Jon
-Original
I understand how to make a dialplan.. that wasn't my question. I
also understand how to make the system add or remove digits from what
the user dialed.
My question is.. if I set it to 'national' doesn't that force the call
to go out with NPA-NXX-? My understanding is if I set it to
on the CLI type this command:
dialplan show [EMAIL PROTECTED]
-and-
dialplan show [EMAIL PROTECTED]
you should see a dialplan returned to you. if not, which is what I
expect, you have to include the section [where6009is] in [local] or
[default]... i.e.
[local]
include = where6009is
On 7/24/07, Asterisk guy [EMAIL PROTECTED] wrote:
-- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold()
(Retry 1)
Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
Jul 24 08:23:17
try priorityjumping=yes in extensions.conf
- Original Message -
From: Michael J. Liberatore [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 23, 2007 11:21 PM
Subject: Re: [asterisk-users] Upgrade Procedure
I was able to get H.323 to work with G729 on 1.2.18. It works real well.
- Original Message -
From: Cesc Santa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 24, 2007 1:54 AM
Subject: [asterisk-users]
--- Noah Miller [EMAIL PROTECTED] wrote:
You can try using ChanIsAvail() to test beforehand
if the zap channels
will accept a call.
I'll try that. Thanks Noah.
My test was with disconnected analog lines.
I will also try to do the same but this time will keep
the line busy by placing a call
read this
http://www.tuxtone.com/index.php/VOIP:Asterisk_Dial_Plan
On 7/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote:
I noticed in 1.4.x I can no longer use n+101 ? I use this all over my
dial plan and wouldn't even know how to replace it. Like when trying to
call out and a
Thanks Tahir. I already got the asterisk-addons though - that's what
I'm having trouble with! BTW - asterisk-addons also provides a
menuselect now. The problem is that the MySQL components all show up
XXX even though I have MySQL installed.
Hugh
On 7/24/07, Tahir Almas [EMAIL PROTECTED]
Is your system compiled as BE or LE ?
bristuff will compile but will not work on BE systems without some
patches/endianess fixes as some of the buffer pointers are little endian
Also with the existing bristuff you will get invalid data due to the fact
that the cache controller of xscale can not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hugolivude wrote:
Thanks Tahir. I already got the asterisk-addons though - that's what
I'm having trouble with! BTW - asterisk-addons also provides a
menuselect now. The problem is that the MySQL components all show up
XXX even though I have
Hi,
please see your ./configure output especially few last lines.
and note missing thins.
Regards
Nasir Iqbal
On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote:
Thanks Tahir. I already got the asterisk-addons though - that's what
I'm having trouble with! BTW - asterisk-addons also
Hello,
I have some problem to start asterisk.
First I have followed a lot of tutorials to complete correctly the install
process. Now it works when I type zttool I can see when I am or not
connected to the PSTN.
But, I run asterisk with verbose and I can't see the call detection.
There is
Any thoughts? Am I stuck with modifying chanspy itself to allow an exit
DTMF?
Message: 6
Date: Thu, 12 Jul 2007 08:43:17 -0400
From: [EMAIL PROTECTED]
Subject: [asterisk-users] exit ChanSpy with DTMF
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]1advertisi
ng.com
There is a theory that says that jitter buffers should not be used until
the end of the voice path where jitter might be introduced. With that in
mind, and in this scenario, the jitter buffers should reside at the ATA
and media gateway;
ATA (SIP UA) ASTERISK NATIVE BRIDGE MEDIA GATEWAY (SIP
To prevent further missunderstanding please do not refer the SI-120 as a
snom
phone. If you need support please contact snom India.
Tim,
If it is sold by snom India, and one is to contact snom India, I can
certainly see how one could infer that it is indeed a snom phone.
John
Hello,
I am experiencing extreme jitter/slowdown on Playback() or Background().
I've looked thoroughly on voip-info.org and elsewhere for help regarding
this issue but cannot figure this out. I can make outgoing calls with no
problems.
I am running Linux 2.6.18-4-686 and Asterisk 1.4.8 and I do
Hi all,
I´ve 2 VoiceTronix 12 cards in my Machine with asterisk 1.4.8 running,
everything is working fine except the detection of far-end hangup.
There is someone in this list with VoiceTronix 12+asterisk and got
working the detection ?
Below are some useful information about drivers
I have one Digium adapter S101I on a local network and I'm losing the
connection periodically.I'm using Asterisk-1.2.21.1
There is no pattern, sometimes it stays connected for a weak sometimes
longer.
In comparison I have few Sipura adapters and they stay connected for
months (never lost
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP
Asterisk SIP ), I get a ring tone. When I
now decide to hang up (e.g. if
nobody answers), the called telephone
Sury, but the phone is no snom phone, it's China trash full of bugs!
snom India is just distributing this trash it.
I can only pointing out this issue, trying to prevent bad user experience
with these phones.
I'm very very unhappy about the situation but it's out ouf my reach to
change this.
Am Montag, den 23.07.2007, 16:21 -0400 schrieb Michael J. Liberatore:
I noticed in 1.4.x I can no longer use n+101 ? I use this all over my
dial plan and wouldn't even know how to replace it. Like when trying to
call out and a channel is busy, would I need to do all if then's??? How
can I
Damon Estep wrote:
Anyone know the answer? Has it been validated with packet captures, or
code review?
All of the timing information should be passed across the bridge in all of the
frames that come in over RTP. I can't say I verified this with packet
captures,
but I did look for this in
Am Montag, den 23.07.2007, 14:33 -0400 schrieb Matt:
Hi,
What dialplan option do I need to send a call out like this:
NPA-NXX- local calls
1-NPA-NXX- - long distance
Won't 'national' send it out NPA-NXX- no matter if it's long
distance or not?
I do not understand your point
On Tue, Jul 24, 2007 at 03:44:29PM +, karim H wrote:
Hello,
I have some problem to start asterisk.
First I have followed a lot of tutorials to complete correctly the install
process. Now it works when I type zttool I can see when I am or not
connected to the PSTN.
But, I run asterisk
Ryan Parlee wrote:
I am experiencing extreme jitter/slowdown on Playback() or Background().
I've looked thoroughly on voip-info.org and elsewhere for help regarding
this issue but cannot figure this out. I can make outgoing calls with no
problems.
When I run zttest I get the following:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Russell Bryant
Sent: Tuesday, July 24, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP jitter buffer and asterisk native
Damon Estep wrote:
Thanks a bunch. So in theory the media gateway at the far end should be
able to properly jitter buffer the entire RTP path from the ATA via
asterisk, correct?
Would this be the same in 1.2 and it 1.4?
Yes, that is correct, but only for 1.4. In the case of Asterisk 1.2,
On Tue, 2007-07-24 at 10:48 -0600, Joseph wrote:
I have one Digium adapter S101I on a local network and I'm losing the
connection periodically.I'm using Asterisk-1.2.21.1
I know there have been some recent fixes to the IAX implementation in
Asterisk to handle a few scenarios when IAX
Hi,
I have installed Asterisk 1.4 with mISDN with the
install-asterisk.tar.gz script from beronet.com. On my system I have two
cards, one a AVM Frit!Card Pci 2.0 and one HFC-S chip. I know both to
work well with mISDN on my system from a previous installation.
Now however, the AVM card works
On Tue, 2007-07-24 at 13:32 -0400, Jared Smith wrote:
On Tue, 2007-07-24 at 10:48 -0600, Joseph wrote:
I have one Digium adapter S101I on a local network and I'm losing the
connection periodically.I'm using Asterisk-1.2.21.1
I know there have been some recent fixes to the IAX
I can only pointing out this issue, trying to prevent bad user experience
with these phones.
I'm very very unhappy about the situation but it's out ouf my reach to
change this.
Yes, and I certainly understand how difficult it can be to get upper
management to understand that their cost
Hi Everyone...
I am running Asterisk 1.2.22 on Debian Etch. I installed it from
sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded
http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c
Dear List;
I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.
I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP
On Tue, 2007-07-24 at 13:27 -0700, bilal ghayyad wrote:
Is it OK? Will it register on my softswitch or will
send call directly without registeration on it?
No, your configuration wouldn't register to the softswitch... but it
will send a call directly to the softswitch. (Registration only tells
Hi,
Install spanDSP 0.0.2-pre26 not 0.0.3.
Le mardi 24 juillet 2007 à 17:02 -0300, [EMAIL PROTECTED] a
écrit :
Hi Everyone...
I am running Asterisk 1.2.22 on Debian Etch. I installed it from
sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded
In your case it will send calls without registering to softswitch . Btw what
does your softswitch expects from asterisk ? like is it configured to
authenticate by username alone , user/pass or ip address ?? People here can
help you better if you post that info .
On 24/07/07, bilal ghayyad
Vieri wrote:
--- Noah Miller [EMAIL PROTECTED] wrote:
You can try using ChanIsAvail() to test beforehand
if the zap channels
will accept a call.
I'll try that. Thanks Noah.
My test was with disconnected analog lines.
I will also try to do the same but this time will keep
the line busy
You cannot detect disconnected analog lines in Asterisk. You can't even
determine of the lines have dialtone. All you can do is determine if
there is a current active asterisk managed call.
Noah Miller wrote:
Hi Vieri -
I'm trying to set a rule to dial out through multiple
Zap groups
Noah Miller wrote:
2) Set callprogress=yes in zapata.conf (if you haven't already done that).
If you set callprogress=yes you will have the opposite problem -- active
calls will be randomly disconnected.
___
--Bandwidth and Colocation Provided by
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's
This should do what you want:
You can call it like this:
exten =
12125551212,1,Macro(dialoutbound,${EXTEN},8005551212,3-Zap/G1/-Zap/G2/-Sip/Nufone/)
The above using the macro below will try zap/g1 first if it's in use
or otherwise unavailable, ti will go to zap/g2 and then sip/nufone.
Hi all,
We have put together a firmware for a range of inexpensive routers.
It has been configured to provide optimum VoIP performance.
We have internally tested it for number of months and it looks very good.
You should be able to run it easily with 20+ phones on local network ( we
still did not
Hi List;
Is there a link that help me to configure asterisk to
register to another gatekeeper as client?
Regards
Bilal
Ready
for the edge of your seat?
Check out tonight's top picks on Yahoo! TV.
hi,
can anyone point me to answering machine beep detection methods or writeups for
*?
thanks,
daveC
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Thanks or all your help!
I've posted the ./configure output below. I noticed that it says:
checking for mysql_init in -lmysqlclient... no
Presumably that's a problem, but I don't know how to fix it!! As I
mentioned, I have MySQL installed and it works fine. rpm -qa
indicates:
Thanks for the reply. Unfortunately that didn't work. What's confusing
is that for the line without any distinctive ring that works correctly
with callerid, the only thing it does is dial the phones, so here's the
entire context:
[add-incoming]
exten = s,1,Dial(SIP/ht1SIP/ht2SIP/gxp1,20)
Hi Mike,
I think that callprogress=yes is right.
but Noah Miller also right.
so the solution will be.
callprogress=yes
busydetect=yes
busycount=4 ; suitable values is above then 4 choose minimum
; as More value more time to wait before hangup.
rxgain=7; you can adjust this
On 7/24/07, hugolivude [EMAIL PROTECTED] wrote:
Thanks or all your help!
I've posted the ./configure output below. I noticed that it says:
checking for mysql_init in -lmysqlclient... no
Presumably that's a problem, but I don't know how to fix it!! As I
mentioned, I have MySQL installed
JR Richardson wrote:
Hi All,
Strange issue, recently I started getting a lot of .lock files in the
voicemail /INBOX folder preventing proper access to voicemail. I can
delete the .lock files and everything is normal. After searching
around, I found some SIP lock file stuff but nothing
Hi dave,
you can use AMD application
for more info please visit
www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
Regards
Nasir Iqbal
On Tue, 2007-07-24 at 17:22 -0400, dave cantera wrote:
hi,
can anyone point me to answering machine beep detection methods or writeups
for *?
Asterisk Project Security Advisory -
++
| Product |
Asterisk |
|+---|
The Asterisk development team has released Asterisk versions 1.2.23 and
1.4.9.
These releases contain bug fixes, including one for a security vulnerability.
The vulnerability is a potential Denial of Service attack when the Asterisk
IAX2 channel driver is configured to allow unauthenticated
Thanks again. Ive posted the rpm query configure.log below. I
noticed that the mysql libraries were in /usr/lib/mysql/ so I tried
./configure --with-mysqlclient=/usr/lib/mysql but end up with the
following:
configure: ***
configure: *** mysql_config was not found in the path you specified:
I tried several and had very poor luck with each I tried. These included
IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I
think Germany that had just changed it's name. All of these had issues. I
could not get Firefly configured at all to talk to Asterisk. Diax, when
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
You can try my IAX2 softphone for windows :
http://www.marccharbonneau.com/asterisk/mediaxphone.php
Hope it fits your need
___
I've had decent luck with PhonerLite, connecting via SIP. The interface
is not the best, but I've been able to connect reliably and make calls.
-Ryan
bilal ghayyad wrote:
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
Joseph wrote:
Thanks, do you know what iaxy firmware version is asterisk 1.4 using?
My asterisk-1.2.21.1 is using iaxy version 23
It is the same version in 1.4, as well.
--
Russell Bryant
Software Engineer
Digium, Inc.
___
--Bandwidth and
Matthew,
I've read through the link and I believe my kernel is setup correctly,
however I am still experiencing this problem.
I do not have a Digium card which this link says may be required, however, I
have no problems with my other Asterisk box running Asterisk 1.2.7.1 and
Linux 2.4.18-3.
Hi Matt,
We have one and it works very well - usual Polycom quality, as others
have attested. The only thing we have noticed is a reluctance to
download its config files via FTP when using a VLAN tag.
CP
Matt wrote:
Hi,
Has anyone here ever used a Polycom IP 4000 Soundstation SIP
Conference
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