Hi
Last month, people reported a crash with Asterisk 1.2.13 and
spandsp-0.0.3 when receiving a fax using fax detection.
Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with
the snapshots for app_rxfax.c and app_txfax.c.
The problem still happens.
Has anyone found how to
Besides that you can use centos-plus repository which has lot of updated
stuff not available in RHEL4 like php5 , mysql5 and all .
On 18/12/06, Carla Schroder [EMAIL PROTECTED] wrote:
On Sunday 17 December 2006 10:47 pm, Andrew Joakimsen wrote:
I've used Asterisk on a bunch of RH 7.3 machines
Or this link :
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
se the /n parameter of Local/ channels.
Cheers
Greg
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
Hi,
I am using CentOS 4.4 [ asterisk-1.2.12.1 ]
I too had problems with RxFax application.
I tried spandsp-0.0.2pre26 spandsp-0.0.3pre23
.0.2 could install, but it crashed
.0.3 doesnt install
Finally I got 0.0.2pre26 running on debian sarge 3.1, without a crash !
- Danny
Jean-Yves
Hi
On 12/18/06, Danny [EMAIL PROTECTED] wrote:
I am using CentOS 4.4 [ asterisk-1.2.12.1 ]
I too had problems with RxFax application.
I tried spandsp-0.0.2pre26 spandsp-0.0.3pre23
.0.2 could install, but it crashed
.0.3 doesnt install
I never had any problems installing spandsp 0.0.2
On Mon, Dec 18, 2006 at 07:17:56PM +1100, Jean-Yves Avenard wrote:
Hi
Last month, people reported a crash with Asterisk 1.2.13 and
spandsp-0.0.3 when receiving a fax using fax detection.
Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with
the snapshots for app_rxfax.c and
Hi,
I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1
link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with
E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and
Hello.
Once again, I came up with a problem for which
I can't seem to find a solution.
I'm not able to make BLF work with Thomson ST2030 phones
and Asterisk (1.2.13).
I've set up hints in dialplan, as well as Subscibe keys
on the phone. The LED status gets updated according to
the associated
Hi
On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Can you provide a backtrace of the crash?
Sure.
I've attached a backtrace for both 1.2.13 and 1.2.14 running the same
version of spandsp and all other libraries.
This is on a Fedora Core 6 machine
(I can not attach the message as it
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone= fr
defaultzone = fr
My Zapata.conf:
[channels]
language=fr
context=from-E1
switchtype = euroisdn
pridialplan = unknown
signalling = pri_cpe
I'm not sure that u have to use a crossover cable. Your telco give u a network
emulation, and u emulate a cpe, so i think u need a straigh cable.
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Noc Phibee
Inviato: lunedì 18 dicembre 2006 12.53
On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Dec 17, 2006 at 12:35:41PM -0600, Kevin P. Fleming wrote:
Samy Antoun wrote:
I noticed that the sound directory is missing from
asterisk-1.4.0-beta4.tar.gz.
This is incorrect; the sounds directory is present and contains two
Hi
what is your operator?
I have some pb on orange...
thx
Noc Phibee a écrit :
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone= fr
defaultzone = fr
My Zapata.conf:
[channels]
language=fr
Hi
it's Colt-Telecom.
you have a TE405P ?
bye
pixiesfr a écrit :
Hi
what is your operator?
I have some pb on orange...
thx
Noc Phibee a écrit :
Hi
anyone have a idea for debug my digium TE405P card ?
My zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=
Hi,
I have Asterisk 1.2.9.1 on a Debian box with a beronet BRI card
(install-misdn-mqueue driver). Sometimes, calls are interrupted by music
on hold without any reason: the caller and the callee are put on hold
for few seconds (they both listen to moh) and then the call is
established
Greetings,
Back in September someone asked about documentation for the new SLA feature
in 1.4, however they received no replies. I thought I might ask the same
question now in December. Apart from sla.conf.sample and a few comments in
app_meetme.c I have been unable to find useful
Hello
that might would be an easy question for someone, but im in doubt
Is there any possibility to pass a call from one asterisk to another and
then to ZAP channel.
For instance
I have
A asterisk with numbering 45670
B asterisk with numbering 45680
second asterisk has TE110P card with single
That would be great if Antony's demand could be satisfied.
Cheers
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DUNDi can do this for you. Advertise the routes you can terminate on Box A.
When you place a call on Box B, have it check your DUNDi cloud, and Box A
will provide the route and terminate the call via zap for you.
Alex
On 12/18/06, Pryakhin Dimitry [EMAIL PROTECTED] wrote:
Hello
that might
I may be making this easier than it is but something like this should
work:
A:
DIAL(IAX2/${ASTERISKB}/[EMAIL PROTECTED])
B:
[context]
exten = EXTEN,1,DIAL(Zap/${EXTEN})
I have this scenario also except we have numerous A servers connecting
via the PRI lines on B servers.
Alberto,
Call pickup is not implemented yet within Thomson ST2030 (1.50 firmware).
More precisely, call pickup current implementation is not Asterisk
compliant.
A new release is scheduled for February (I've got this confirmed by Thomson
10 minutes ago) but we don't know if call pickup will be
I'm happy to report that with a very litte change to app_devstate.c
(just in the way ast_device_state_changed_literal() is called)
that module just compiles and works fine even without bristuffing
anything.
BTW I'm using a Thomson ST2030S phone with a status key subscribed
to a DS/xxx hint.
Perfect.
I guess I could not find it on my own because I was searching for a
variable, but a function is fine with me!
Thank you,
Damon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper
Sent: Sunday, December 17, 2006
Any idea what actually causes this problem? Is it an error with the
zaptel programs or asterisk? Or does this problem lay with the telco
providers. Seems odd that if you restart the driver, everything is good
again (mine does as well). This leads me to think its either asterisk
being unable to
On 12/18/06, Anthony Kava [EMAIL PROTECTED] wrote:
Greetings,
Back in September someone asked about documentation for the new SLA
feature
in 1.4, however they received no replies. I thought I might ask the same
question now in December. Apart from sla.conf.sample and a few comments
in
Is this fixable? It seems as though the channels aren't clearing up
after use, and after 2 or 3 incoming calls, i get the fast busy. if I
wait a while, or if i restart zaptel, the channels clear up again.
Any ideas?
Thanks,
Rob
Henry.L.Coleman wrote:
Sounds like you have a disconnect
Here's what I have, it's to early for me to think so hopefully looking
at mine helps :D
extensions.conf:
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701
sip.conf:
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
yusuf wrote:
Hi,
I just got hold on an Orion E1 30 port GSM Gateway, and I am having
problems trying to get the E1 link to come up. I am using Asteisk
1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both
the Digium and Samgoma types, as I have successfully hooked up to many
Hi
I've got a script like this
exten =
s,1,SetVar(CALLFILENAME=/var/www/recordings/${TIMESTAMP:0:8:7}/${UNIQUEID})
exten = s,2,AGI(recordstart.py,${ARG1},${CALLERIDNUM},${CALLFILENAME},Ind)
exten = s,3,DIAL(ZAP/g2/${ARG1},70)
exten =
On 12/7/06, nik600 [EMAIL PROTECTED] wrote:
I am experiencing this:
1 - A,B,C are SIP users logged on QUEUEA with ringall strategy
2 - I call QUEUEA
3 - A,B,C start ringing
4 - nobody answer
5 - D logs on the QUEUEA
6 - D doen's receive any call, but A,B,C are still ringing
How can i avoid
On Sat, 2006-12-02 at 02:07 -0500, Doug Crompton wrote:
I am running an old SUSE 7.3 system, 2.4 kernel and glibc 2.2
I picked up the ncurses-devel rpm and it now requires glibc 2.3
I found a glibc 2.4 rpm but I am a little reluctent to install it. It
would be a disaster to lose this system.
Leo Ann Boon wrote:
yusuf wrote:
Hi,
I just got hold on an Orion E1 30 port GSM Gateway, and I am having
problems trying to get the E1 link to come up. I am using Asteisk
1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both
the Digium and Samgoma types, as I have
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi list.
Has anyone used any commercial or open source application to integrate
Asterisk into MS Outlook 2003 which can be used to place calls directly
to
On Sun, 2006-12-17 at 13:51 -0600, Chris Johnson wrote:
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extesions.conf:
[globals]
polycom430=SIP/101
[internal]
;exten = 101,1,Dial(SIP/101,10,)
;exten = 101,2,VoiceMail([EMAIL PROTECTED] )
;exten =
I think Thirdlane has a software plugin for Asterisk that does this.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
Forgot linkage
http://www.thirdlane.com/outlookdialer.htm
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim -
Yikes! Thanks for that disturbing info.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Joakimsen
Sent: Friday, December 15, 2006 10:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Good Commercial Grade Service
Thanks! I will check them out
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of LST
Sent: Friday, December 15, 2006 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Good Commercial Grade Service Provider?
On
Hi list.
Has anyone used any commercial or open source application to
integrate Asterisk into MS Outlook 2003 which can be used to place
calls directly to contacts from Outlook?
And if so how well does it work?
Here you go... Enjoy:)
http://www.bicomsystems.com/products/C/P/319/288/
You're right. I just untarred asterisk-1.4.0-beta4.tar.gz. The sounds folder
is there, but it is empty except for Makefile and sounds.xml. I am not expert,
but when I looked at the Makefile, it appears that it prompts the user to pick
a format for the sounds files (ulaw, wav, etc), and then
I've never used it but...
http://www.snapanumber.com/
Looks ok feature-wise - plus there's a free version to take for a test
drive.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Soderblom
Sent: 18 December 2006 14:46
To:
We are using the snap program in both Outlook 2003 and 2007, it also handle
click to dial from all Microsoft office apps, FireFox, thunderbird and I
believe Internet explorer. Check it out at
http://www.snapanumber.com
On 12/18/06, Senad Jordanovic [EMAIL PROTECTED] wrote:
Hi list.
Has
Please do not take this as a flame against cyberdyne-ip.com. That is
not the intention. I am just wondering how businesses like this expect
to stick around when they are charging rates this low.
You can find a whole list of other providers that thought this model
would work at:
I am thinking of turning on the NAT option in our Cisco phones (and the
corresponding sip.conf modification) to allow the phones to be taken outside
the LAN.
Can anyone think of any reason not to just always turn on the NAT enabled
option? I can't think of a reason not to always operate these
I am experience repeated digits when connecting a call from SIP using any codex
I have tried the same things to fix this.
If anyone knows why please let me know.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX) 1-866-638-1254
For Information on PBX
While I don't see anything wrong with this, I'm no expert. I took my
instructions from the following URL and they worked fine... I have
the subscribecontext in General and it works fine. What is the
firmware on the GXP? old firmware may be related
-t-
we probably need to ask in dev- list, because seems that only developers
knows, how to use/test SLA feature ;-)
Anthony Kava wrote:
Greetings,
Back in September someone asked about documentation for the new SLA feature
in 1.4, however they received no replies. I thought I might ask the
a few weeks ago I encountered the same problem.
I found out that asterisk is crashing when app_rxfax.so is calling line
327 of app_rxfax.c 'ast_frfree(int);' out of the testing tree running
with actual spandsp-0.0.3
commenting this line out it doesn't crash *, but that's no solution
it do work
Hi.
How do I cause voicemails that land in one mailbox to be delivered to
another?
I.e. I have a incoming call extension that rings all the phones. If it
times out, the caller drops into the general mailbox. I would like messages
dropped in the general mailbox to fall into another users
On Monday 18 December 2006 9:24 am, Ejay Hire wrote:
Hi.
How do I cause voicemails that land in one mailbox to be delivered to
another?
I.e. I have a incoming call extension that rings all the phones. If it
times out, the caller drops into the general mailbox. I would like
messages
Resending as message didn't show up the first time
I need to access MySQL from the dial plan. Currently I am using the MYSQL
function:
exten = *78,n,MYSQL(Connect asterisklocal localhost asteriskuser password
asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
I was wonder if anyone is rumming this combination of hardware:
Colomachine.com: CM62
Digium Card: TE405P
I need a rackmount to send to a data center and this combination fits my
budget. Has anyone else used colomachine with asterisk? how has it
performed? I plan to run the latest
Well I don't see anything that specifically states digium.. but I do
see this. which would be a problem if this is the digium card..
04:04.0 Ethernet controller: Unknown device d161:2400 (rev 11)
IRQ 3
05:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721
IRQ 3
On 12/15/06,
Hello,
We have several clients with GXP2000 in their network and behind NAT. We
have one particular client that has several GXP2000 behind a Linksys RV082
VPN Firewall/Router which is doing NAT services. According to SIP packet
inspection, it detects it's a symmetric NAT.
The problem we have is
Hi,
Is there a way I can stop logging this specific messages:
Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due
to usage limit of 1
Without having to completely stop logging all error messages in my log
files.
Thanks,
Remi
Ciao kjcsb,
I need to access MySQL from the dial plan. Currently I am using the
MYSQL function:
exten = *78,n,MYSQL(Connect asterisklocal localhost asteriskuser
password asterisk)
exten = *78,n,MYSQL(Query resultid ${asterisklocal} CALL\
Hi yusuf,
I am working right now on a similar setup.
If its the PRI type theres not so much on the syncing part. You need
the PRI crossover rj45, theres info on voip-info on that and Orion has
software to configure via Serial cable the E1 PRI as NET/USER and Time
syncs.
I setup mine via
On Mon, Dec 18, 2006 at 01:55:37PM -0500, Remi Quezada wrote:
Hi,
Is there a way I can stop logging this specific messages:
Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due
to usage limit of 1
Without having to completely stop logging all error messages in my
Do you have STUN Enabled? I had similar when I had STUN turned on. I
found it better to turn off stun and place in sip.conf nat=route.
Also use NAT Keep-Alive on the ATA that is NAT Timeout on the Router.
Good Luck,
Mark Coccimiglio
IS Director
Payroll Services Hawaii, Inc.
I'm not sure that any solution with the MySQL dialplan command is going to be
ideal. You also can't nest your queries, ie the connectid/result id seems to
only be good for one resultset at a time... try doing something like
findme/followme with that!
Doug.
-Original Message-
From:
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote:
I would be very surprised if your modem is supported by Asterisk - but I
suppose it's worth a try.
What does 'zap show status' and 'zap show channels' show in the Asterisk
CLI?
PaulH
OK. I got the Motorola X100P put in:
Relevant
On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote:
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote:
I would be very surprised if your modem is supported by Asterisk - but I
suppose it's worth a try.
What does 'zap show status' and 'zap show channels' show in the
On Mon, 2006-12-18 at 22:19 +0200, Tzafrir Cohen wrote:
On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote:
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote:
I would be very surprised if your modem is supported by Asterisk - but I
suppose it's worth a try.
What
I changed
fxsks=2
to
fxsks=1
and now ztcfg works:
camille ~ # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
camille ~ #
I bought the card on ebay. The seller sent some configuration
On Mon, Dec 18, 2006 at 02:29:20PM -0600, Michael Sullivan wrote:
On Mon, 2006-12-18 at 22:19 +0200, Tzafrir Cohen wrote:
On Mon, Dec 18, 2006 at 02:10:23PM -0600, Michael Sullivan wrote:
On Wed, 2006-12-13 at 14:21 +1100, Paul Hales wrote:
I would be very surprised if your modem is
when placing calls to the system through SIP, I got these messages,
Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type 'Zap'
I am having trouble setting this system up and wonder if some one help me.
Does anyone know what is missing if anything to get 2 phones on my
asterisk home server to be able to call each other.
I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2
extensions 5060/5061, this is
On Mon, Dec 18, 2006 at 11:22:13PM +0200, O.Kamal wrote:
when placing calls to the system through SIP, I got these messages,
Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 19 00:26:55 NOTICE[5570]:
Why do I need g729 license?, i am not doing any transcoding in the middle.
it is all g729 passthrough.
softphone---asterisk---zap
On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 18, 2006 at 11:22:13PM +0200, O.Kamal wrote:
when placing calls to the system through SIP, I got
O.Kamal wrote:
Why do I need g729 license?, i am not doing any transcoding in the
middle. it is all g729 passthrough.
softphone---asterisk---zap
I believe you are. Zap is ulaw.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
What zap device do you have that encodes/decodes g729?
- Original Message -
From: O.Kamal
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, December 18, 2006 4:37 PM
Subject: Re: [asterisk-users] ZAP problem
Why do I need g729 license?, i am not
Here's where I stand:
camille asterisk # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
camille*CLI zap show status
Description Alarms IRQbpviol
CRC4
On Mon, Dec 18, 2006 at 03:53:55PM -0600, Michael Sullivan wrote:
Here's where I stand:
camille asterisk # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
camille*CLI zap show status
On Tue, 2006-12-19 at 00:09 +0200, Tzafrir Cohen wrote:
On Mon, Dec 18, 2006 at 03:53:55PM -0600, Michael Sullivan wrote:
Here's where I stand:
camille asterisk # ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default)
I'm not sure that any solution with the MySQL dialplan command is going to
be ideal. You also can't nest your queries, ie the connectid/result id
seems to only be good for one resultset at a time... try doing something
like findme/followme with that!
Thanks
What is a better way to do it
i have digium TDM2404E, I was thinking that zap devices are not related to
any kind of codecs. I will try setting my soft phone and asterisk server to
use ulaw, to see how things will go...
On 12/18/06, Mailing List [EMAIL PROTECTED] wrote:
What zap device do you have that encodes/decodes
Hello Asterisk Users,
I guess the subject says the most of it; here goes some more
detail:
- Running Asterisk 1.2.14
- Objective: record all calls managed by a specific queue
- Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}
Facts:
- If the UNIQUEID chan var is used in the
On 12/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
a few weeks ago I encountered the same problem.
I found out that asterisk is crashing when app_rxfax.so is calling line 327
of app_rxfax.c 'ast_frfree(int);' out of the testing tree running with
actual spandsp-0.0.3
commenting this line
Hi
On 12/18/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hi
it's Colt-Telecom.
you have a TE405P ?
you don't mention what's wrong with it though...
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asterisk-users mailing list
To UNSUBSCRIBE
I have the following setup:
- UAs registered with SER/OpenSER
- SIP peers (non cached), extensions, voicemail setup (not message storage)
defined in Asterisk 1.2 using Realtime
When a message is left in the user's mailbox, no Notify message is sent to
SER.
1. If the SIP peer is defined in
Did I forget to mention I had STUN enabled? :)
Well, that did it. Your suggestion worked perfectly.
Does anyone know what a reasonable NAT Keep-Alive to use, if you don't
have access to their firewall/router configuration?
Thanks,
Daniel
-Original Message-
From: Mark Coccimiglio [EMAIL
I have the following setup:
- UAs registered with SER/OpenSER
- SIP peers (non cached), extensions, voicemail setup (not message
storage) defined in Asterisk 1.2 using Realtime
When a message is left in the user's mailbox, no Notify message is sent to
SER.
1. If the SIP peer is defined in
Well, I am making some progress. I have made some changes as defined below
and now have a green line on the BLF, but it still does not indicate when
the extension receives a call or goes off hook.
Here are the changes:
the [ext-local-custom] context no longer exists
the subscribecontext in
What is a better way to do it then in terms of performance, security,
and
flexibility? Using exec and a shell script, or agi or something else?
Setup extconfig to have realtime access to the database/table you want to
pull info from, then in the dialplan use the app Realtime.
-= Info
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor
I prefer to keep my NAT Timeouts short ( ~5 minutes) and lets the
applications be responsible for keeping the connections open. **Most**
consumer grade routers use a timeout interval of 1 hour to 1 day. A
safe figure to start with is 600 seconds (10 minutes) and see if anyone
complains.
Could the fact that asterisk isn't aswering the phone be a firewall
issue? What port(s) on TCP and UDP do I need to open for incoming calls
to be allowed to go to asterisk?
___
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asterisk-users
Hi,
How could I possibly inform incoming callers that the number they'd dialed is
monitored and recorded.
I wanted that when a call-in or call-out is made, a playback will be played to
inform caller callee that thier line is monitored prior to start conversation.
Thanks.
Angel
camille*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoincomingen
This should show something like this :
panoramix*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudofrom-pstn en
The call for papers for the upcoming Hack in The Box Security Conference
2007 - Dubai is now open.
HITBSecConf2007 - Dubai will take place at The Sheraton Creek hotel and
will run from the 2nd till the 5th of April 2007. Keynote speakers for
the conference will be Mikko Hypponen (Chief
I have asterisk running in a wrt54gs attached is a pap2 with 2
extensions working on it, the problem now is that there is lots of echo,
some rythm in the background, and the voice is delayed by about 4 or 5
sec's between the 2 extensions. memory usage is about 15 to 20 megs so I
think I can
Setup extconfig to have realtime access to the database/table you want to
pull info from, then in the dialplan use the app Realtime.
Thanks. I didn't know that you could use RealTime in the dialplan like that.
I thought is was just for sip, extensions etc.
I created a wiki page at
I was wondering if anyone had any experience getting a 7960+7914 working
with any of the chan_sccp modules. I've got a 7960G with 6.0(5.0) and a
factory fresh 7960G with 3.1(MF.G2). I've got 2 7914s fresh out of the box
brand new. I hook them up and all I get is red lights on all of the
The problem I am running into is that when the call to my cellphone is made,
it appears as though the call completes so it never rolls to asterisk
voicemail.
Here is my current config:
exten = 102,1,Dial(${sipura},10,)
exten = 102,n,playback(pls-wait-connect-call)
exten =
On Mon, 2006-12-18 at 22:03 -0500, Time Bandit wrote:
camille*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoincomingen
This should show something like this :
panoramix*CLI zap show channels
Chan Extension Context
With the playback command?
I think we are missing something here.
PaulH
On Mon, 2006-12-18 at 19:01 -0800, Angel Heart wrote:
Hi,
How could I possibly inform incoming callers that the number they'd
dialed is monitored and recorded.
I wanted that when a call-in or call-out is made, a
Is the problem just when you don¹t answer the cell phone? Many cell phones
go to a voice announcement when they¹re turned off or not answered, and
Asterisk thinks the call has been answered. The other issue could be that
your gateway (asterisk1) is answering the call before the outbound leg is
exten = s,1,Answer
exten =
s,n,Set(REC=${URIENCODE(${STRFTIME(,America/Toronto,%Y%m%d-%H%M%S)}-${CALLER
ID(number)}-TESTBOARD-${UNIQUEID})})
exten = s,n,MixMonitor(${REC}.wav)
exten = s,n,Playback(this-call-may-be-monitored-or-recorded)
Note that I intentionally start the recording BEFORE
. Could it hurt something when they are used inside our LAN with NAT
enabled?
The answer is no!
With my test bed, I found that Asterisk can detect Endpoint behind NAT(match
via and src_ip).
So, once the EP is on LAN (same side of NAT) then they work as if there is
no NAT. The option of nat=yes
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a
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