https://issues.asterisk.org/jira/browse/18998
https://issues.asterisk.org/jira/browse/18998 may have the answer,
particularly the patch bug18998-1.8.2.3.diff.txt
Alec
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
2011/6/15 A J Stiles asterisk_l...@earthshod.co.uk
On Wednesday 15 Jun 2011, Olivier wrote:
At the moment, my console is full with messages such as :
[Jun 15 09:06:25] WARNING[2140]: chan_dahdi.c:3369 pri_find_dchan: No
D-channels available! Using Primary channel 9 as D-channel anyway!
Hi,
I try to solve my problem with asterisk and BLF function.
I have registered peers from realtime with subscriptions but only type
is mwi (shown by 'sip show subscriptions').
Peers are registered from behind the NAT - may it be the cuase why
they not subscribed with dialog-info+xml?
Regards,
Hi,
You may used the Page() function of asterisk. Which will work the same as
you are required at this moment.
On Wed, Jun 15, 2011 at 12:51 PM, Alex Balashov
abalas...@evaristesys.comwrote:
On 06/15/2011 01:34 AM, Nikhil wrote:
Hi
Asterisk support dialout conference?.My requirement is
On Tue, 14 Jun 2011, Florent THOMAS wrote:
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011 17:54, Gordon Henderson a écrit :
On Sat, 11 Jun 2011, Florent THOMAS wrote:
Hy all of you,
Is anybody has a tutorial for integrate a siemens gigaset as180 and
connect it to Asterisk.
Le 15/06/2011 10:53, Gordon Henderson a écrit :
On Tue, 14 Jun 2011, Florent THOMAS wrote:
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011 17:54, Gordon Henderson a écrit :
On Sat, 11 Jun 2011, Florent THOMAS wrote:
Hy all of you,
Is anybody has a tutorial for integrate a
Thanks for the helps
I use channel originate command to achieve this.
Command:
asteriskCLI channel originate SIP/201 application ConfBrigde 1234
This will make a call to the 201 user and when connected,it will be
routed to conference room .
Thanks
NIkhil
On 06/15/2011 02:17 PM, virendra
i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL.
WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane,
On 15 Jun 2011, at 11:20, mahesh katta wrote:
i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL.
WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG
A lot of filesystems are case sensitive. Maybe you wrote your configuration in
caps? This would also explain why you
Is there a dahdi_cfg in your boot sequence? When I modify dahdi config
files I always launch dahdi_cfg otherwise I get errors like yours.
Giorgio
On 06/14/2011 05:37 PM, Olivier wrote:
After a reboot, I can't reproduce the problem anymore which is quite
frustating.
2011/6/14 Tzafrir Cohen
Hi,
I think you need to update *waittime* parameter in .call file please put
atleast 10 seconds.
for more understanding please try to read
*http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out*
Regards
Dhaval
On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic
Sir,
thanks for reply .
exten = 8501,1,VoicemailMain(s${CALLERIDNUM})
exten = 8501,2,Hangup
exten = 4578909,1,AGI(agi://127.0.0.1:4577/call_log)
exten =
4578909,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten =
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it UP, I have to configure
the two ports and both of those two ports to take the timing from span 1.
Why this, I do not know ! Although I am using only one E1 connected to span 1,
so why I
Hi all,
I am using asterisk1.2(vicidial). I am using like pbx . In this how can I
confugure the internal conference calls. suppose I have A,B,C,D,E users
these all peoples should be internal conferece . for them i was give
101,102,103,104,105 extensions. For this scenario what can I do exact
Hi,
I Required digium PRI cards, single span, dual span, quad core .
so any body give me cotaion for this cards and I required also
grandstream fxs/fxo devices . give me for this quotation .
price and details..
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
On Tue, Jun 14, 2011 at 11:36:33PM +0200, Olivier wrote:
Hi,
My genconf_parameter is :
# grep -v ^# genconf_parameters
lc_countryfr
context_linesremote
group_lines1
bri_sig_stylebri
echo_canoslec
pri_termtype
SPAN/*NT
(I also
HI list,
no idea?? :)
M.
On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.comwrote:
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE
What company card you have? Copy paste your dahdi config and
chan_dahdi.conf
--
Sent from my iPhone
On Jun 15, 2011, at 6:53 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
The problem was related to something else.
The Digium card has two PRI ports, actually to get it UP, I have to
I'm on 1.8.3.3 and it does the same thing. Once you log back in it says you
have a message. You press 1 to play and she just says First then gives you
options to delete, save etc. The message is in the INBOX as msg0001.wav
currently.
From: Alec Davis
Sent: Wed 6/15/2011 4:12 AM
To:
2011/6/15 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Jun 14, 2011 at 11:36:33PM +0200, Olivier wrote:
Hi,
My genconf_parameter is :
# grep -v ^# genconf_parameters
lc_countryfr
context_linesremote
group_lines1
bri_sig_stylebri
echo_can
This was a bug in 1.4, 1.6.x, and 1.8. It is fixed in the latest release of
each of the Asterisk versions. Check the Changelog for 1.8.4, you might see
the bugtracker ID with the patch.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The same issue was present in 1.6 a few weeks ago and is fixed in latest
1.6. Maybe latest 1.8.4 does not have this issue.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Wednesday, June 15, 2011 8:44 AM
Is there an easy way to setup diaplan so when someone pushes a digit such as
* during a call, they will be transferred to another destination.
For example, a caller is hearing ringing while calling a UA, but instead of
waiting for the UA to pick up, they can push * and go directly to that UA's
The latest 1.8.x solved the problem for us on multiple servers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 15, 2011 9:29 AM
To: 'Asterisk Users Mailing List -
Hi,
it seems to be fixed in 1.8.4. At least I can't reproduce it there.
Karsten
Am Mittwoch, den 15.06.2011, 09:29 -0400 schrieb Mike:
The same issue was present in 1.6 a few weeks ago and is fixed in
latest 1.6. Maybe latest 1.8.4 does not have this issue.
Mike
From:
Hello !
i am new to this list and asterisk.
I run asterisk 1.4 on a OpenSuSE 11.4. My SIP Provider needs my IP to
connect the local area number to my IP and also for there firewall.
I plan to run asterisk in a pacemaker cluster that is not the problem
and works.
My problem is the virtual IP
Le 15/06/2011 11:06, Florent THOMAS a écrit :
Le 15/06/2011 10:53, Gordon Henderson a écrit :
On Tue, 14 Jun 2011, Florent THOMAS wrote:
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011 17:54, Gordon Henderson a écrit :
On Sat, 11 Jun 2011, Florent THOMAS wrote:
Hy all of
Hi,
If your cluster's virtual IP is using ip aliasing (eg eth0:0), i
think your problem come from UDP flows, they are, in opposition to
TCP flows, unconnected, so the IP stack take the shortest
route/interface to send them, wich is when this is the default
The card is Digium card T2XXP (PCI) as I mentioned in my email.
I added the configuration for the second port (span 2) to work, otherwise it
does not work.
I just added the below lines in the files system.conf and chan_dahdi.conf, all
other lines are the default lines. The asterisk version is:
On Wed, 15 Jun 2011, Florent THOMAS wrote:
Le 15/06/2011 11:06, Florent THOMAS a écrit :
Le 15/06/2011 10:53, Gordon Henderson a écrit :
On Tue, 14 Jun 2011, Florent THOMAS wrote:
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011 17:54, Gordon Henderson a écrit :
On Sat, 11 Jun
Le 15/06/2011 21:14, Gordon Henderson a écrit :
On Wed, 15 Jun 2011, Florent THOMAS wrote:
Le 15/06/2011 11:06, Florent THOMAS a écrit :
Le 15/06/2011 10:53, Gordon Henderson a écrit :
On Tue, 14 Jun 2011, Florent THOMAS wrote:
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011
On Wed, 15 Jun 2011, Florent THOMAS wrote:
Do you know some devices that aren't so locked?
None of them are locked by default - it's only the service providers that
lock them into their own networks - so if you buy anything from an online
supplier that
doesn't
[snip very hard to follow thread]
Linksys devices are locked at the factory AFAIK and cannot be
unlocked. If a Linksys ATA is what you are after, you want a model
that ends with '-NA'.
j
Thanks for answering.
I wasn't looking for a linkSys, I inherit of the device that my customer
own
Great,
Thanks to all of you for leading me to a solution.
regards
Le 15/06/2011 21:45, Florent THOMAS a écrit :
[snip very hard to follow thread]
Linksys devices are locked at the factory AFAIK and cannot be
unlocked. If a Linksys ATA is what you are after, you want a model
that ends
Hello,
Yes, the issue I am having is currently only with Google Talk. Wonder
if what development will be made to fix this issue.
--Elliot
On Wed, Jun 15, 2011 at 9:20 AM, Vladimir Mikhelson v...@mikhelson.com wrote:
Elliot,
I do not think Issue # 17993 is related. As Terry Wilson says on
On 06/15/2011 04:40 PM, Elliot Murdock wrote:
Hello,
Yes, the issue I am having is currently only with Google Talk. Wonder
if what development will be made to fix this issue.
At some point it will be fixed, and then Google will break it again.
Google Talk/Google Voice connections to
You should probably grab a free DID as a failover from gtalk. Have gvoice ring
them both and answer the one that comes through first. In my tests. I have
better luck with the DID than with gtalk.
-- cobra2
Http://linuxindixie.info
Kevin P. Fleming kpflem...@digium.com wrote:
On 06/15/2011
Dears;
OK, I start beleive that the problem in the TFTP and the files that I placed
there.
Now, I am using the Phone as skinny, and the files that are placed in the
directory /var/lib/tftpboot/ as following:
CTLSEPB8BEBF22AB62.tlv
SEPB8BEBF22AB62.cnf.xml
XMLDefault.cnf.xml
Well, actually the
Hi,
I am a question to handle incoming goggle voice. I have put several GV
accounts into the jabber.conf. How can I direct different accounts to
different extensions?
Help with example is much appreciate
Thanks,
CK
--
_
--
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.
Any suggestions?
CK
--
exten = accou...@gmail.com,1,Answer()
exten = accou...@gmail.com,n,Wait(2)
exten = accou...@gmail.com,n,SendDTMF(1)
exten = accou...@gmail.com,n,Dial(SIP/device1)
exten = accou...@gmail.com,1,Answer()
exten = accou...@gmail.com,n,Wait(2)
exten = accou...@gmail.com,n,SendDTMF(1)
exten =
Thanks and will try.
On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
exten = accou...@gmail.com,1,Answer()
exten = accou...@gmail.com,n,Wait(2)
exten = accou...@gmail.com,n,SendDTMF(1)
exten = accou...@gmail.com,n,Dial(SIP/device1)
exten =
Thanks a lot for all your comments.
Finally I have figured out the problem by looking into source code.
If callcounter=yes and notification is enabled for ringing or hold in sip.conf
file, asterisk queue will not fork the new incoming call to the members already
in ringing or inuse state.
Hi
Price for Digium (4) span digital T1/E1/J1/PRI PCI card = Rs. 56,000.00 +
5% VAT / 5.00% VAT (Delhi)
Price for Digium (4) span digital T1/E1/J1/PRI PCI card with Echo = Rs.
87,000.00 + 5% VAT / 5.00% VAT (Delhi)
Price for Sangoma (4) span digital T1/E1/J1/PRI PCI card = Rs.
Can you provide me express cards price also
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone
Hi,
Yes you can use Dial(sip/xxx,30,Ttr) option then it will transfer to any
where you want.
On Wed, Jun 15, 2011 at 7:03 PM, vip killa vipki...@gmail.com wrote:
Is there an easy way to setup diaplan so when someone pushes a digit such
as * during a call, they will be transferred to
Hi List,
I want to secure my server from the hacker's. What is the case by which I
can protest it.
I have done security of Dialplan, Sip,IAX base security. For linux we are
working on Iptables. What else is left so that I will do it too...
--
-
Thanks and regards
Virendra Bhati
I thought the idea was that Asterisk Engineers already know the
answers to such questions?
On 06/16/2011 01:52 AM, virendra bhati wrote:
Hi List,
I want to secure my server from the hacker's. What is the case by
which I can protest it.
I have done security of Dialplan, Sip,IAX base security.
Well, I ran a simple test by trying to configure the second port to use the DNS
SRV record, as described below.
Here is what I have: (sanitized)
==
Proxy_2_ diehlnet.com /Proxy_2_
Outbound_Proxy_2_ fqdn /Outbound_Proxy_2_
Display_Name_2_
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