Hi All
I am looking for a small scale Email to fax solution
Searches seem to throw up
AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to
http://www.noojee.com.au/products/noojee-fax/fax-overview/
email12fax http://wpkg.org/email2fax/index.php/Main_Page
I would
Hi,
I'm having an issue with all my calls going out my SIP provider. I'm using
a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way -
it's great and actively growing).
I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP -
real IP addr. is
Hello list,
I have configured extconfig.conf to save queue log into my MySQL-DB.
I notice however that there is still logging too in
/var/log/asterisk/queue_log.
How can I disable queue logging into text files ?
Kind regards,
Jonas.
--
Hi all,
I try to figure out why I have empty :
sip show subscriptions
list in may asterisk 1.6.
When device is registering to asterisk I can see in log:
NOTICE[25603]: chan_sip.c:21518 handle_request_subscribe: Received SIP
subscribe for peer without mailbox: 1010
but
sip show subscriptions
Set queue_log = no in logger.conf. By default it is set to 'yes'.
[SATISH]
On Wed, Jun 8, 2011 at 12:30 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
I have configured extconfig.conf to save queue log into my MySQL-DB.
I notice however that there is still logging too in
On 06/08/2011 09:10 AM, Satish Barot wrote:
Set queue_log = no in logger.conf. By default it is set to 'yes'.
[SATISH]
Will there then still be queue logging at all ?
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation
Give it a shot and check! :)
Yes you will have your Queue log records in table.
[SATISH]
On Wed, Jun 8, 2011 at 12:46 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 06/08/2011 09:10 AM, Satish Barot wrote:
Set queue_log = no in logger.conf. By default it is set to 'yes'.
[SATISH]
Thanks Paul,
Link was too awesome. I read and check all related command too.
Thank you for your help.
On Wed, Jun 8, 2011 at 2:37 AM, Paul Belanger pabelan...@digium.com wrote:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any
HI Krishna,
As per your suggestion I have changed Makefile of appKonference. Which is
listed below.
And after that I have reinstalled same module again.
# turn app_konference dtmf on of off ( 0 == OFF, 1 == ON )
DTMF = 1
Now* how I know that DTMF is activated and working ? Is these any option
satish patel wrote:
We are getting hangup cause 18
http://networking.ringofsaturn.com/Routers/isdncausecodes.php
*Cause No. 18 - no user responding.*
This cause is used when a called party does not respond to a call
establishment message with either an alerting or connect indication
within
Hi Pan Dhaval,
In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based
call center with our flexqueue application for icson.com. It has the below
features,
1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two
are failover configured with heartbeat and
Thanks for reply,
But I'm able to call those number from my cell phone and othere pri.
I'm only having this issue on 2 pri line rest are working ?
--
Sent from my iPhone
On Jun 8, 2011, at 5:44 AM, Doug Lytle supp...@drdos.info wrote:
satish patel wrote:
We are getting hangup cause 18
Hi,
Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
$ sox foo.wav -e stat
just a remark for people using newer(?)/other version
Hi,
I am using CentOS 5.6 and I am getting error message
In my case old command is find.
On Wed, Jun 8, 2011 at 5:25 PM, Karsten Wemheuer k...@gmx.de wrote:
Hi,
Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
For the record, it seems to be a SIP-ALG issue. It's fixed now.
Vieri
--- On Wed, 6/8/11, Vieri rentor...@yahoo.com wrote:
Hi,
I'm having an issue with all my calls going out my SIP
provider. I'm using
a softphone registering to a local Asterisk PBX (I'm using
Jitsi by the way - it's
Ist the same operator connected to the pri-line? Perhaps another
telco-operator can not connect to the desired destination - for whatever
reason.
Am 08.06.2011 12:55, schrieb Satish Patel:
Thanks for reply,
But I'm able to call those number from my cell phone and othere pri.
I'm only having
Ist the same operator connected to the pri-line? Perhaps another
telco-operator can not connect to the desired destination - for whatever
reason.
Am 08.06.2011 12:55, schrieb Satish Patel:
Thanks for reply,
But I'm able to call those number from my cell phone and othere pri.
I'm only having
We have two sites. BOSTON and California
We are having only issue with California PRI line related cause 18 but BOSTON
pri has no issue. All settings are same on both Asterisk. Today i will talk to
service provider and will see.
pridialplan=uknown fixed many issues except cause 18
-S
Hi ALL,
After upgrade 1.8 my MWI wasn't working I do have setting in voicemail.conf.
Do i need to do anything else to fix my MWI on polycom 501 ? It was working
with 1.2 asterisk.
pollmailboxes=yes
--
On 11-06-07 10:20 PM, Jose P. Espinal wrote:
Hello Guys,
After the Wiki was updated to the 3.5.X version, my username is no loger
available:
user: khratos
mail: j...@slackware-es.com
I had some documents on my personal space. Is there a way to recover the
account?
Yes,
My account is also
All major changes are listed in the UPGRADE.txt files included in the 1.8
tarball.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Wednesday, June 08, 2011 9:57 AM
To: asterisk-users
Truly speaking, I went though that file and i found nothing in that file
related major changes. It was working perfect before 1.2
May be i am missing some configuration option. Do you know any debug method to
make it work ?
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
On 06/08/2011 01:09 AM, Paddy Grice wrote:
Hi All
I am looking for a small scale Email to fax solution
Searches seem to throw up
AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to
http://www.noojee.com.au/products/noojee-fax/fax-overview/
email12fax
Following is my debug and look like its not sending MWI NOTIFY message to phone
Reliably Transmitting (no NAT) to 172.30.245.143:5060:
OPTIONS sip:7623@172.30.245.143 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.46:5060;branch=z9hG4bK5bd640a3
Max-Forwards: 70
From: asterisk
Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make sure
your mailboxes specify a voicemail context on each mailbox= line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
I do have that
sip.conf
[7623](cam-exten)
callerid=Satish Patel 7623
accountcode=Satish Patel
mailbox=7623@default
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
Date: Wed, 8 Jun 2011 11:03:24 -0400
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
Starting on line
Is 7623 listed in voicemail.conf under the [default] section?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Wednesday, June 08, 2011 11:15 AM
To: asterisk-users
Subject: Re:
Do you think i should enable ?
; searchcontexts=yes
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
Date: Wed, 8 Jun 2011 11:03:24 -0400
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
Starting on line 147 of UPGRADE-1.2.txt in the latest 1.8 tarball. Make
sure
Yes its under [defailt] section at voicemail.conf
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
Date: Wed, 8 Jun 2011 11:17:26 -0400
Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI
Is 7623 listed in voicemail.conf under the [default] section?
-Original
If call comes into PBX-A and based on the DNIS it comes into my box PBX-B
my box then says ring phone C. Person answers. They want to transfer the
call
to a phone going back out PBX-A. All this is fine of course.
my question is when phone C transfers the call is there a way PBX-B can
drop out
I assume you misspelled default in your e-mail and not voicemail.conf. If
not, that is your problem.
When there is a new message in a mailbox, does voicemail show users show new
messages for that mailbox?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Yes its under [defailt] section at voicemail.conf
Sorry it my typo error.
When there is a new message in a mailbox, does voicemail show users show
new messages for that mailbox?
Yes, I can see there are 10 voicemail
root@campbx1:~# asterisk -rx 'voicemail show users' | grep -i 7623
Hi List,
I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW.
Currently I'm running Asterisk 1.8.3.3.
I have the following problem, if I do the update to the actual 1.8.4.2.
There are several commands on the CLI which are not working or even not
present like
core show
Interesting thing is when i reload sip.conf i got MWI lamp working on polycom
501
But its not working when anyone leave voicemail. Do you know its some timeout
or polling setting in sip.conf ?
Still my question is my my asterisk not sending NOTIFY message ? Do i need to
subscribe my
On 06/08/2011 01:09 AM, Paddy Grice wrote:
Hi All
I am looking for a small scale Email to fax solution
Searches seem to throw up
AsterFax http://sourceforge.net/projects/asterfax/ which seems to go
to http://www.noojee.com.au/products/noojee-fax/fax-overview/
email12fax
What do you think to do with the solution ? Cause we developed it ourselves and
is in run on more company, if you want I can talk you about it.
In any case, Avantfax I remember to be a frontend for Hilafax.
I don't know the other one, sorry
Enrico Cicconi
www.rdmnet.it
Cordialmente
Enrico
On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
Interesting thing is when i reload sip.conf i got MWI lamp working on
polycom 501
But its not working when anyone leave voicemail. Do you know its some
timeout or polling setting in sip.conf ?
Still my question is my my
Hey Guys!
Please help me to find out issue. I have two PRI
## Span 1: WPT1/0 wanpipe1 card 0
span=1,1,0,esf,b8zs
bchan=1-23
hardhdlc=24
echocanceller=mg2,1-23
## Span 2: WPT1/1 wanpipe2 card 1
span=2,2,0,esf,b8zs
bchan=25-47
hardhdlc=48
echocanceller=mg2,25-47
Sometime my calls got through
Hello list,
can anyone tell me if this card :
http://www.audiocodes.com/product/ipm-260-sip
is compatible with Asterisk (DAHDI) for use as PCI PRI card ?
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by
This card is a standalone SIP media server on a PCI blade. But you can make
it work with Asterisk for that you have to tweak Asterisk source and as well
as you have to buy API from audiocodes if I am not wrong.
Instead of this why can't you use Sangoma or Digium cards?
On Wed, Jun 8, 2011 at
Bad day today. Why this new JIRA system not working. I have created issue and
submit and i got blank page.. Please someone help me to create BUG!!!
--
_
-- Bandwidth and
Hi List,
I have working experience of asterisk with PRI lines. Recently I have took
VoIP routes from my provider. My basic issue is that now how asterisk will
behave in such case. I mean in PRI call will come as below process
PRI - - Digium Card - - Dadhi/Zap - - Extensions.conf
What will be
Hi List,
I am making outgoing call from asterisk to GSM network with the help of VoIP
trunk(SIP trunk) then I am not geting any caller ID at destination end. Is
this the asterisk issue or VoIP trunk issue?
Is this is due to asterisk then how we solve it? I already user
You mean this one?
https://issues.asterisk.org/jira/browse/ASTERISK-17984
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Wednesday, June 08, 2011 2:17 PM
To: asterisk-users
Subject: [asterisk-users]
Hi List,
When we make calls into asterisk with the help of our mobile, landline
number, Cisco 79XX series then we didn't able to here any IVR which is
playing into asterisk server. But when we dial from SIP softphone then all
is working fine and we are able to here the IVR sound files.
What is
On Wed, 8 Jun 2011, virendra bhati wrote:
I have working experience of asterisk with PRI lines. Recently I have
took VoIP routes from my provider. My basic issue is that now how
asterisk will behave in such case. I mean in PRI call will come as below
process
PRI - - Digium Card - -
On Thu, 9 Jun 2011, virendra bhati wrote:
When we make calls into asterisk with the help of our mobile, landline
number, Cisco 79XX series then we didn't able to here any IVR which is
playing into asterisk server. But when we dial from SIP softphone then
all is working fine and we are able to
I get this on my Mac:
Safari can’t open the page.
Safari can’t open the page
“https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t
establish a secure connection to the server “issues.asterisk.org”.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
A number of people are reporting that Safari is not working properly with JIRA.
Use Firefox or Chrome for now.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW- Huntsville, AL 35806 - USA
www.digium.com -=- www.asterisk.org -=-
On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant russ...@digium.com wrote:
A number of people are reporting that Safari is not working properly with
JIRA. Use Firefox or Chrome for now.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW
On 11-06-08 10:34 AM, Paul Belanger wrote:
On 11-06-07 10:20 PM, Jose P. Espinal wrote:
Hello Guys,
After the Wiki was updated to the 3.5.X version, my username is no loger
available:
user: khratos
mail: j...@slackware-es.com
I had some documents on my personal space. Is there a way to
On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel satish...@hotmail.com wrote:
It not working on iPhone. It's saying not able to make secure connection
--
Sent from my iPhone
Satish, Can you share what the SSL/TLS Cert says? Safari and mobile
platforms have a smaller list of CAs, just to make
Hi,
We have two pri line and I want to see how asterisk distribute
outgoing call per channels
I meant it use first last channel 47 or it will use first channel?
Or it will allocate dynamically ?
--
Sent from my iPhone
--
We have two pri line and I want to see how asterisk distribute
outgoing call per channels
I meant it use first last channel 47 or it will use first channel?
Or it will allocate dynamically ?
Extracted from chan_dahdi.c:
Dial(DAHDI/pseudo[/extension[/options]])
Can a quad or six core server with 4 GIG RAM running asterisk 1.4
handle 1000 polycom phones.
jerry
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
If I click on the link below, without jira, Safari goes to here:
https://issues.asterisk.org/main_page.php
And yes it works.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 8, 2011, at 1:54 PM, Kevin P. Fleming wrote:
On 06/08/2011 02:27 PM, Andrew Latham
Do you mind checking again? I'm now able to access my account again.
Yes, everything is Ok. now, even my documents on personal space.
--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs
--
Hello all,
We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups
which previously worked fine have stopped working.
Can anyone advise if there has been a change in how pickups work?
Here is an example where 1000101 is trying to pick up a call to 1000103:
Awesome!!
Do you know if i want to use only specific channel for call out then how do i
write dialplan ? I want to use channel 25 specific for my extension
DAHDI/25/ or DAHDI/i2/25/XXX
Date: Wed, 8 Jun 2011 17:25:44 -0500
From: rmudg...@digium.com
To:
I hope my understanding is not wrong!
(1) DAHDI/i2/25/XXX, is not a valid format for Dial. Rather it
should be DAHDI/i2/XXX and it would use a channel from span 2
(/etc/dahdi/system.conf) for outgoing call.
(2) To dial from channel 25 , use DAHDI/25/XXX
[SATISH]
On
Hi Virendra,
It may be problem for rtp packet port forwarding if u can dial through DID
number.
You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port.
please, write how can you dial call mobile or other devices. e.g. DID
number, PRI number etc.
--
Best Regards,
Rajnikant
Hi Steve,
Thanks for reply. Is this method will follow on DID incoming calls too?
I mean when we call on DID then call will come to my server and then I want
to move this call to any SIP extension. But call will not come to extension
just got message *device not in use. *But device already
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