Why did you set rx/txgain to 0.5 ? Most people use 0.8, but I
use no gain by setting it to 1.0, which works here good.
Does anyone know if you need to set rx/txgain to 0.0 to disable gain... or
it is a percent value...
DV
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Hi,
my Asterisk records CDR logs in a MySQL table.
Is there anyone having a SQL query to find max load (max concurrent calls)
of my system?
Thanks in advance
--
Domenico Viggiani
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I have set pickupgroup and callgroup for zap, sip and iax2 devices.
Everything is working good with zap and sip and between these two.
Iax2 pickupgroup and callgroup seems to be broken. I cannot
pickup a call to IAX2 from SIP.
Is there somewhere a bug ?
I am running: Asterisk 1.2.9.1
I can confirm this.
AMP/TrixBox is a wonderful project but if you like to tweak
something or you became a moreexperienced user, it will became
soonas a straitjacket.
I'm still struggling to clean AMP config files to work with
a plain Asterisk install.
From: [EMAIL PROTECTED]
Just installed!
Use 6.1.1 (beta) because 6.1 has a few of registration problems.
Bye
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Koopmann, Jan-Peter
Sent: Friday, June 23, 2006 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I'm doing some experiments with SIP vs. IAX2 trunks as an alternative to
PSTN and I noticed that, if voip link is down, failover to PSTN is almost
immediate with the SIP trunk and VEEERY slow with the IAX trunk.
Is there a specific reason? Some timeout to set in iax.conf?
Thanks
--
Domenico
Hi,
since last august, I worked on Asterisk and PRI lines, making a good
experience.
Now I need some information about ISDN BRI: I know that there is no native
support in Asterisk and I need some third-part driver. Then I read often
about different cards (Junghanss, Eicon, Beronet, etc), different
And once again I am reminded why I shouldn't bother helping
people here.
Not even a 'thanks'.
Thanks againm no problem, but I think that:
1) I already thank you in my previous message:
(Upgrade to freePBX 2.1.1, it's much better, really)
I upgraded to custom, 'made with vi' files,
Hi,
after a few of upgrades, I noticed these messages in full debug log:
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
Jun 21 12:58:11
Does anyone know why this row:
exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3)
generate this error:
ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE,
expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
!=
^
?
I was unable
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
Hi,
after a few of upgrades, I noticed these messages in full debug log:
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
switchtype Jun 21
No, IMHO does it appear when you issue a reload command on
the CLI. Because this options need a complete *-restart.
Yes, they appears when I issue a reload.
I will check if there are also when I restart.
Thanks
DV
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From: Rob Thomas
That's freePBX or AMP code that we've since fixed - The
replacement line is
exten =
s,2,GotoIf($[${CALLERID(name):0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3) ; check for old prefix
Yes, ok. I'm gradually fixing all the code using Asterisk 1.2 syntax.
(Upgrade to
From: Rob Thomas
Looks like you've stopped compiling libpri. All those options
that are being ignored, are being ignored because they're for
PRI, and you don't have PRI support in zaptel.
Uh?
If I don't have PRI support in zaptel, how are my 80 employees calling their
homes now?!
:-)
DV
Did you see this after a reload?
Asterisk will ignore some settings when doing a reload.
Only a restart will pickup changes to the settings mentioned
in your mail.
True. In fact, after a restart, I don't see any WARNING.
Thanks
DV
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I'm getting some of these errors:
ERROR[7244] chan_sip.c: Got SUBSCRIBE for extensions without hint. Please
add hint to 464 in context from-internal
All these extensions are not local (sip) but other analog phones attached to
a legacy PBX downstream.
Any idea?
Thanks
--
Domenico Viggiani
Hi,
I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it
doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to
click Re-register in the web interface.
I set:
- Support broken Registrar: On
- RTP Encryption: Off
Any help?
--
Domenico Viggiani
I noticed that the waiting message indicator does not lit
when I have a message in my voice mail. Any suggestion
why this is happening?
Double check:
- mailbox=ext@context for your extension in sip.conf
- if your 'home directory' is under /var/spool/asterisk/voicemail/context
context is usually
Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Festival RPM?
um, yum install festival worked for me.
-Original Message-
From: Mimmus [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 9:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial
If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works
well with 1.1.0.11 firmware.
I can send you this firmware, if you mail me off-list.
Bye
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
You need to encode txt configuration file using tool provided on GS site.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthias Fechner
Sent: Wednesday, June 14, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi,
calling a partner on the other side of a SIP trunk, call gets disconnected
immediately after answer. This is the content of log file:
Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel:
SIP/cerved-out-6eba
Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging
Hi,
using Sangoma drivers:
- doing 'lsmod', I see:
zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
I'd like to avoid loading all these modules. What have I to do?
- do I need to have 'zaptel' startup script under /etc/init.d ?
Thanks
--
Domenico Viggiani
to another.
Thanks a million in advance.
Mimmus wrote:
I used this approach to gradually migrate from a legacy Alcatel PBX:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel PBX At
first, Asterisk did nothing, only passing calls to/from Alcatel.
Then I started to use a bunch
Hi,
is there a RHEL4 RPM for the Festival text-to-speech system?
Thanks
--
Domenico Viggiani
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I used this approach to gradually migrate from a legacy Alcatel PBX:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel PBX
At first, Asterisk did nothing, only passing calls to/from Alcatel.
Then I started to use a bunch of SIP phones directly connected to Asterisk.
Now I have the great
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
On Thursday 08 June 2006 23:08, Callum McGillivray wrote:
We would like to add an analogue card, plug
our fax machine in and have Asterisk simply detect the fax
and pass it
through to the fax machine
Hi,
often on this list I read about transcoding as the heaviest activity for an
Asterisk server, together with high IRQ rate (especially with Digium
cards...).
Is there a way to monitor if Asterisk is engaged (by mistake or by design)
in transcoding or any other heavy activity?
Or a checklist to
good to known.
I played with the idea to buy one of these.
Unacceptably bad voice quality. Point.
You would suggest GrandStream then?
Surely better in my experience.
DV
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Asterisk-Users
Grandstream phones and it was a real jump: users now are happy,
I had only one RMA, quality and stability are good and I'm able to focus
myself on improving Asterisk features.
IMHO: +100 Euros for a phone are a theft!
--
Mimmus
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* Mimmus [EMAIL PROTECTED] [07-06-06 16:52]:
At first, I tried some chinese phones (AtCom) and they were
a disaster.
you talking ybout this phone?
http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2
Yes
DV
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Hi,
I just migrated my Asterisk installation from 1.2.1 to another server with
1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
All is OK but now I'm not able to see MWI indication for new messages on all
my Grandstream GXP2000 phones
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Thomas Kenyon
Mimmus wrote:
Hi,
I just migrated my Asterisk installation from 1.2.1 to
another server
with 1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory
Hi,
how can I convert .wav files to .WAV:
# file greet.*
greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
using 'sox'?
Thanks
--
Domenico Viggiani
Hi,
as already said in others messages on this list, I'm rewriting my dialplan
using AMP/FreePBX as starting point.
I saw that AMP/FreePBX uses the concept of USERS/DEVICES, quite interesting
but not useful to me now. It defines USERS/DEVICES association in AstDB and
then uses dialparties.agi
Hi,
is there an Asterisk app (or AGI script) to look up names in a LDAP
directory?
--
Domenico Viggiani
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Hi,
during gradual migration to Asterisk, I put Asterisk in front of a legacy
Alcatel PBX:
PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX
After successful deployment of VoIP phones, it's time to drop Alcatel PBX!
I'd like to keep some of analog lines to support modem, fax and some older
stuff.
Olivier Krief [EMAIL PROTECTED] wrote:
PS: How many users were at start connected to Alcatel PBX ?
What did you do for voicemail during migration?
I had ~110 extensions.
During migration, I simply avoid to give Asterisk goodies to Alcatel users.
Every extension migrated to VoIP could be
like it, you'd probably be better off writing your own
dialplan or alternatively, rewrite it's entire functionality outside of an
agi and then submit the mod to freepbx to streamline freepbx more.
p
From: Mimmus [EMAIL PROTECTED]
To: 'Asterisk Users Mailing
I know only IAXmodem, a software doing what you want using IAX protocol.
Some guys on this list use it with some success.
Mimmo Viggiani
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giorgio Incantalupo
Sent: Thursday, May 25, 2006 10:10 AM
Title: VLAN info
We are currently using VLANs for all our
networks.
Setuping a VoIP VLAN was simply a matter of configuring
some switches. Defining higher 802.1p priority for switch ports on this VLAN was
the following logical step.
We don't
use VLAN tag on the phones directly.
DV
.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
mohamed kerbachi
Sent: Thursday, May 25, 2006 3:51 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] VLAN info
Hi Mimmus,
Can you give us some exemples about Defining
astdb, etc.
p
From:
"Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List -
Non-Commercial
Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May
2006 10:21:46 +0200Subject: RE: [Asterisk-Users]
macro-dialHi,I digged in dialparties.agi and
Hi,
I'm trying to edit an AMP-derived dialplan: the macro dial uses the AGI
script dialparties.agi to find the extension to call.
I'd like to drop this script: does anyone can explain me what is its main
job?
Thanks
--
Domenico Viggiani
___
Subject: Re: [Asterisk-Users] Plan to free myself from AAH
On 5/17/06, Mimmus [EMAIL PROTECTED] wrote:
I was thinking to this plan:
- install another server with Red Hat 4 U3
- install PHP, MySQL and other usefuls stuffs
- download latest version of Asterisk and third parts
AMP dialplan is full of garbage and perpaphs is not fully 1.2
compatible but it is anyway an Asterisk, working dialplan!
As example:
May 17 18:35:40 WARNING[8625] app_db.c: This application has been
deprecated, please use the ${DB(family/key)} function instead.
May 17 18:35:40 WARNING[8625]
Thanks for the advice. indications.conf is now existent and
Asterisk is reloaded but the problem still persists.
Reloaded? Peraphs restarted is better...
DV
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To
Colin was right!
I forgot that AMP dialplan makes intensive use of database keys as AMPUSER
and DEVICE: I understand its logic but migration is postponed after a GREAT
dialplan rewriting :-(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mimmus
First, you can remove the quotes aorund your variable
reference. I've seen examples with it, but you don't need
it.
I'm not sure: if variable is empty, you got an error.
In addition, double quotes around text that may contain spaces
will force the surrounded text to be evaluated as a
Hi,
I'm actually using a slightly old version of AAH with Asterisk 1.2.1,
because at first install it was perfect for my moderate knowledge of
Asterisk. It is working well but I gradually introduced many changes to
dialplan during normal use and now I'm feeling like in a straitjacket!
Moreover I'd
Hi,
I'm trying to match a few of numbers in a GotoIf; numbers are not starting
with but contain some strings:
GotoIf($[${CALLERIDNUM} =~ 984836|984899|498993|644110]?8:11)
Expression result is always '0'.
Where am I wrong?
Domenico Viggiani
___
Hi,
I know that this is a more strictly FOP related question than Asterisk but
I'd like to know if regexp buttons support a '-' char, i.e.:
[_Zap/1-.*]
...
In fact, I have:
Zap/1 to Zap/10 as incoming channels
Zap/11 to Zap/15, Zap/17 to Zap/21 as outgoing channels
(it is an E1 PRI)
and I'd
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Louis-David Mitterrand
Why is IAX2 so flaky?
If I did not resort to SIP for inter-asterisk communications
I would be out of a job at this time.
I fully confirm! I moved to SIP for an
Hi,
I have '*1' in my features.conf file and I'm facing with a serious problem:
- A and B are engaged in a call
- C and D are engaged in a different call and decide to record their
conversation hitting *1
- at the end, A and B are able to see C/D call recording using ARI with
their user/pwd!!!
Hi,
I have ~25 AtCom AT320 phones (PA1888S based) and they was not a good
experience for me: I ran the risk to abort my Asterisk project thanks to
them!
I tried both with SIP and IAX2 firmware of every version but voice quality
was often unacceptable. In addition, they lose often registration and
Any help?
# gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE
-DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o
app_wakeme.o app_wakeme.c
app_wakeme.c: In function `app_exec':
app_wakeme.c:101:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Iedema
1.2.7.1 is your problem. I've updated the site with a 1.2 compatible
version as well as squashed a stupid bug in v0.1.0. At the moment I
don't have access to a 1.2 test box so let
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Iedema
...
You're
still using the incorrect version. Download the 1.2 version
and let me know...
OK, solved. Thank you.
DV
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Hi,
what's thereal use of cfg.txt file during Grandstream GXP2000 provisioning?
Thanks
--
Domenico Viggiani
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Hi,
I'm using RPMs from http://www.laimbock.com/asterisk/ and they works well
(thanks to the author!). They include some patches to provide additional
functionalities.
Now I'm trying to re-create the original compiling environment to recompile
some other apps (app_pickup2, app_ldap, etc) and I
Asterisk::LDAP is unrelated to app_ldap.
Just pre-install openldap-devel for your distro, download app_ldap.c, put it
under apps dir of Asterisk source tree and recompile Asterisk normally.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Joao
on different nets
as my * box with no problem.
- Waldo
On Apr 28, 2006, at 11:12 AM, Mimmus wrote:
Hi,
I have a lot of GXP-2000 phones not registering with
Asterisk server.
After two days of attempts, it seems that problem is due to
the fact
that phones and server are not on the sme
Hi,
during tests, I configured different SIP accounts on the same phone.
Now I see this 'sip show peers output':
Name/username HostDyn Nat ACL Port Status
259/25910.97.1.19 D 5060 OK (8 ms)
232/23210.97.1.19 D 5060 OK (7 ms)
where
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Aaron Daniel
Are you using Realtime or static sip.conf?
Static (AsteriskAtHome base setup)
Thanks
DV
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Hi,
I have a lot of GXP-2000 phones not registering with Asterisk server.
After two days of attempts, it seems that problem is due to the fact that
phones and server are not on the sme network.
Do you know if this is known issue?
--
Domenico Viggiani
Hi,
is there a way to completely disable TFTP/HTTP provisioning on the
Grandstream GXP-2000?
Thanks
--
Domenico Viggiani
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Waldo Rubinstein
Make sure dtmf-mode is set to rfc2833 in both sip.conf as
well as in the GXP-2000.
OK, thanks, but problem is more general: often phone doesn't send even a
packet to the Asterisk
Hi,
I'm planning to install a new Asterisk server with a Digium TE410P card.
Can I use Red Hat Advanced Server 4 (latest update)?
Is this a good choice?
Is recompiling Asterisk simple with kernel 2.6?
Thanks
--
Domenico Viggiani
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Perpahs you need a Wait(2) before Answer() in the dialplan, because telco
send CallerIDName after some time (second ring, I suppose).
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jonathan k. Creasy
Sent: Wednesday, April 19, 2006 3:28
Do your (wonderful) RPMs install also on CentOS?
I suppose so because it is a Red Hat clone...
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Axel Thimm
rpms for Fedora Core 1-5, RHEL 3-4 and RHL 7.3-9 have been updated
OK, your solution is fine but I'd like a more generic solution to adapt it
to my current [EMAIL PROTECTED] setup.
Thanks anyway
Mimmus
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter J Dean
We do it slightly different, rather than multiple macros, we
do it within
I have now:
exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
;exten = _9011Z.,103,Dial(SIP/011${EXTEN:[EMAIL PROTECTED])
exten = _9011Z.,104,NoOp(${DIALSTATUS})
I configured two trunks for my outgoing calls:
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]
maybe firewall tends to close iax connection, you can try to
decrease qualify check interval (maybe qualify=5000?) PJ
Peraphs. 'qualify = 1000' seems to alleviate the problem.
Thanks
Domenico
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maybe firewall tends to close iax connection, you can try to
decrease qualify check interval (maybe qualify=5000?) PJ
Peraphs. 'qualify = 1000' seems to alleviate the problem.
Thanks
Domenico
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how can I put fax server functionality on Asterisk? * as a
reliable fax server for 500-1000 fax/day (mostly incoming)?
Fax server should be like HylaFax, i.e. stable, low
maintenance and functionality like receiving fax as email
with PDF attachment, sending faxes per WHFC.
Asterisk doesn't
Hi,
I have same setup:
PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel 4400 PBX
with some IP phones directly connected to Asterisk and a lot of
analog/digital phones connected to 4400.
When I call from an IP phone to an Alcatel one, I'm able to see full
CallerIDName.
I set it using:
I don't know if I'm using Q.Sig or EuroISDN!
1) it's in config file
2) Should be easy to check when you say what kind of PABX
card you use:
PRA/PRA2/BRA2 - EuroISDN
DLT - qsig
OK, I'm using EuroISDN.
Thanks
DV
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Pavel Jezek wrote:
I have same problem, do you have asterisk box behind nat?
No, they are not behind NAT, peraphs there is a Checkpoint firewall.
Bob McDowell wrote:
It's been a while, but I didn't think those two terms were
necessarily exclusive. Checkpoint firewalls can provide
Hi,
I have a IAX2 trunk between two sites (connected with an high bandwidth
link) but sometime/often I get:
chan_iax2.c: Auto-congesting call due to slow response
and call is dropped (and routed on a PSTN link).
In iax.conf, I have:
[iax-out]
username=iax-in
type=peer
trunk=yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Pavel Jezek
I have same problem, do you have asterisk box behind nat?
No, they are not behind NAT, peraphs there is a Checkpoint firewall.
DV
___
My question is: how can I set specific caller id for outgoing
PRI calls?
Here in Italy I have a E1 PRI line with DID: +39 local-zone-prefix
did-block-prefixdid-block-ext
I was able to set CallerIDnum only after some attempts: I had to set it only
to:
did-block-prefixdid-block-ext
without using
Sorry for delay.
I never tried personally but received this recipe from the author of the
first one in this wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold
Keep me infomed if works.
--
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto
,Wait(5)
exten = 6,103,Goto(6,1)
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
exten = 9,2,Hangup
exten = i,1,Goto(outside,s,1)
exten = t,1,Goto(outside,s,1)
exten = T,1,Goto(outside,s,1)
Don't blam eme if there is some error.
--
Mimmus
On 3/22/06, Mimmus [EMAIL PROTECTED] wrote:
What are these error messages?
Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Failed to read
gains: Invalid
argument Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Failed to
read gains:
Invalid argument Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Updated
EuroISDN uses uLaw, so Asterisk does as well, because it
doesn't need to do transcoding then...
Sure? At my knowledge, in Europe aLaw is always used.
Am I wrong?
Thanks again?
Mimmus
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and all calls use aLaw.
--
Mimmus
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on 2, with 0
conference users
Mar 22 17:41:10 VERBOSE[3592] logger.c: -- Registered channel 2, PRI
Signalling signalling
...
(for all channels)
I have a PRI E1 line configured with:
txgain=0.0
rxgain=0.0
in zapata.conf.
--
Mimmus
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, not here!).
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Mimmus
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Unfortunately in Italy doesn't work: Italy and Spain uses Protocol Type2 and
app_SMS doesn't support it (to my knowledge).
http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePS
TN.pdf
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users
And don't forget to set callgroup/pickupgroup to
each one in your sip.conf
Call pickup works among IAX phones?
Mimmus
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any
ringing phone within the callgroup number (eg, 2 in this example).
Does this call pickup work with IAX2?
If yes, how, if there is no callgroup/pickupgroup setting in iax.conf?
More in general: does call pickup work between different protocols?
Thanks
Mimmus
PickUp2:
http://linux.thorsten-knabe.de/asterisk/pickup.jsp
works very well.
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Panton
Sent: Monday, March 20, 2006 4:50 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non
Hi,
I'm unable to pickup a call (*8) directed to a SIP phone from a IAX2 phone.
Is it normal?
I don't see ant pickupgroup/callgroup setting in iax.conf...
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Domenico Viggiani
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Your manager interface is not so bad: simple but working as
a charm.
I love ncurses
interfaces!
A goodexercise of programming with Asterisk Manager
API.
Keep up the good work.
Mimmus
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Look also at AudioFrames setting on your phone.
I read that it needs to match 20ms packet size of Asterisk packets and it
depends from codec you use.
Mimmus
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From http://nerdvittles.com/:
Max Channels Bug Remains. A bug has been reported because of a deprecated
command that makes [EMAIL PROTECTED]'s calculation of maximum channels invalid.
To fix it, goto AMP-Maintenance-Config
Edit-extensions.conf-macro-dialout-trunk and comment out line s,7 so that
Can someone explain me attended transfer with Grandstream GXP-2000?
Hitting TRNF button, I get:
Dial number (BLIND) or
Select line (ATTENDED)
What's the exact meaning of 'Select line'?
Thanks
Mimmus
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with both XXXext and YYYext
In other words, dialplan is shared between servers.
Actually, we have two Alcatel PBX 4400 working in this way.
Can I do this with Asterisk?
Thanks
Mimmus
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Asterisk
Sorry for my ignorance but what are 'HINTS'?
Thanks
Mimmus
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