with Asterisk.
Regards.
On Tue, 6 Dec 2016, 1:40 p.m. Toshaan Bharvani | VanTosh, <
tosh...@vantosh.com> wrote:
On 05/12/16 17:57, Gopalakrishnan N wrote:
> True agree, problem is somehow the people purchased am supporting to
> overcome that. Trying level best... around 20 pho
TrueAgree.
:)
On Mon, Dec 5, 2016 at 11:37 PM wrote:
> > True agree, problem is somehow the people purchased am
> > supporting to overcome that. Trying level best... around 20
> > phones has been purchased
>
> Ah, yes, the "we purchased
True agree, problem is somehow the people purchased am supporting to
overcome that. Trying level best... around 20 phones has been
purchased
On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal,
wrote:
> With all the money you plan to invest in firmware,
t; I previously had experience of upgrading the Cisco build to the SIP build
> on Cisco 7641 handsets, which have 2 similar builds, but none of the
> techniques seemed to apply this time around.
>
> Cheers,
> Steve
>
>
> On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N <gopalakr
gt; they work very much like the Cisco SPA handsets.
>
> I also ended up with a non-3PCC handset and it is useless, and as far as I
> can tell they cannot be re-flashed.
>
> Cheers,
> Steve
>
>
>
> On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N <gopalakrishnan...@gma
Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the
phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
upload woth TFTP due to some reason it's getting failed. Do I need to load
3pcc firmware or anyway to Configure from the phone itself or from the
Hi,
I have cisco 8841 IP phone. could someone light up how to configure with
Asterisk.
Thanks in advance.
Regards,
Gopal .
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk
Finally got it worked, the issue was E164 callerid format, where i set it
up, after removing the E164 format its was thru.
Regards
On Fri, Feb 12, 2016 at 9:31 PM Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Now incoming works fine, this is because of my SonicWALL firmw
Now incoming works fine, this is because of my SonicWALL firmware issue,
tried with different SonicWALL inbound works.
But for outbound am getting 408 request time out error in the NAT on VPN
tunnel.
On Fri, Feb 12, 2016 at 3:50 AM Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
&g
Hi,
I came thru ISDN UUI (User-User Information) protocol which is defined in
this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt
But I don't understand how to use this with Asterisk. Any idea would be
much appreciated.
Thanks.
Gopal.
--
:* asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnan N
gopalakrishnan...@gmail.com
*Sent:* Monday, August 18, 2014 4:13 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] log caller hangup
It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss)
Regards
On Fri, Aug 15, 2014 at 6:20 AM, CDR vene...@gmail.com wrote:
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Hi,
You can use Hangup handler. May be this post can you help you,
http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html
Regards
On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg p...@greenberg.pro wrote:
All,
I would like to log a message whenever a party hangs up a
Can we have concurrent calls via asterisk manager interface, lets say
around 1000 or 1000+ concurrent calls.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Thanks Johan. Are you using this application for any credit card processing?
On Fri, Apr 4, 2014 at 5:29 PM, Johan Wilfer li...@jttech.se wrote:
2014-04-03 18:58, Gopalakrishnan N skrev:
Hi,
Anybody using PAGI scripts,
http://marcelog.github.io/articles/pagi_tutorial_create_
Hi,
Anybody using PAGI scripts,
http://marcelog.github.io/articles/pagi_tutorial_create_voip_telephony_application_for_asterisk_with_agi_and_php.html
Would like to know the feasibility to build a IVR solutions.
Regards
--
_
--
Am looking for a service provider who can provide enterprise SIP trunk with
100 channels concurrent sessions.
I see some like Inphonex, Broadvoice... and etc
Is there any suggestions for the service providers.
Regards
--
_
SIP options message is due to check the peer registration is keepalive. As
per my understanding it might be because of network flap may be wireshark
trace can give you any clue.
Regards
On 13 Feb 2014 23:41, Tim Nelson tnel...@rockbochs.com wrote:
Greetings-
I recently experienced an odd
Enable debugging module and backtrace and re-compile so that you will
bactrace of the crash logs.
Regards
On 14 Feb 2014 10:29, Arun Ram arunram@gmail.com wrote:
Hi guys,
I need a desperate help from you regarding this asterisk crash issue.
On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram
Which is the best way around to integrate Asterisk with VoiceXML like
VoiceGlue...! Am using Asterisk 11.2.1.
Regards.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Anyone using Voiceglue with latest Asterisk 11.6 certified version?
On Mon, Jun 20, 2011 at 10:00 PM, Jean-Denis Girard jd.gir...@sysnux.pfwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Le 20/06/2011 04:40, Gopal krishnan a écrit :
Have anybody integrated
OpenVXI
Hope basically depends on the codec Asterisk will playback the file
automatically
On 23 Jan 2014 19:25, Gareth Blades mailinglist+aster...@dns99.co.uk
wrote:
On 23/01/14 13:38, Ishfaq Malik wrote:
Hi
Is there any way to change the preferred audio playback format in
asterisk (I'm using
protocol. Which Perl AGI library
are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
Sent: Saturday, November 30, 2013 1:27 AM
To: Asterisk Users Mailing List - Non-Commercial
Thanks... I got it working actually I found with this command /usr/bin/perl
-d agi file name from this I got to know that my library is missing and
installed Asterisk-perl module and now its fine.
Once again thank you.
On Mon, Dec 2, 2013 at 3:05 PM, Gopalakrishnan N
gopalakrishnan
Alao enable cel table that will have all the information
On 29 Nov 2013 23:25, Todd R. tjrl...@live.com wrote:
I do this by writing custom CDR. I write the agents extension write into
the CDR records. This makes is easy to just parse through the CDR and get
all the info you need about the
I have a Perl AGI script updating some values to database like recorded
file path, unique ID and callerid. When I run the script with test
dialplan, its not updating to database.
Whereas database connection is fine, when I run agi debug I see only Tx
packets not Rx packets, firewall is also OFF.
I have a situation where Asterisk is not releasing the channel for Attended
transfer immediately once I transferred and hangup from my side. The call
is still ongoing and disconnecting after the third party disconnected.
I see that its bug in the Asterisk, but not sure its fixed in version
If you are getting like this dropped packets then nothing to worry.. thisis
just an cli message in my case I face this but there is no voice delay
in actual call.
On 22 Nov 2013 21:11, Eric Wieling ewiel...@nyigc.com wrote:
Are you getting errors like this?
[Nov 22 10:39:36]
You can have something like this,
exten = _,1, Answer
exten = _, 2, voicemail ($EXTEN)
On 25 Sep 2013 05:04, Tim Nelson tnel...@rockbochs.com wrote:
Greetings-
I have an odd scenario where I need to dial an extension (lets call it
555), the system prompts for a list of voicemail
What does this mean of bad magic internal error, SIP to SIP calling is
fine, when I use SIP via GSM I have this, and asterisk restarts
automatically. Asterisk version which am using is 11.1.2.
Regards
--
_
-- Bandwidth and
Hi,
How much CPU utilization will it take when I use G729 transcoding via
hardware based transcoder.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
caller to different
queue from this context.
--Satish Barot
Ahmedabad, India.
+919978599700
On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Will Keypress option will work when am in the queue and hearing MoH?
Lets say a caller is waiting
also if am not wrong queue timeout will also applicable for this.. !
On Wed, Aug 28, 2013 at 11:37 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
oh great thanks...
On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot
satish4aster...@gmail.comwrote:
Yes you can. Check the 'context
Hi,
Will Keypress option will work when am in the queue and hearing MoH?
Lets say a caller is waiting in queue and while he is hearing MoH, can he
key in some DTMF and go to some other queue? is that possible?
Regards
--
_
--
You can use AMI Commands and run sip set debug from that you have to
capture the response code.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command
Regards,
On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer mkaga...@gmail.comwrote:
B.H.
Hello, i'm using AMI Originate
Basically I have some background noise like keyboard stoke or clicking
sound in random basis, I need to measure that, when I check my IPLC its
fine, and with my Telco service provider its fine...
So am trying to conclude with some solution... trying to identify the root
cause.
Any advice would
Hi,
Can Ingress and Egress can be used in Asterisk, so that Jitter can be
calculated...!
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
Even I tried the type as friend.. but no use...
On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
, Gopalakrishnan N wrote:
Even I tried the type as friend.. but no use...
On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret
on server 1
On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:
Even I tried the type as friend.. but no use...
On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02
Hi,
Am getting dead or silence calls at sometimes for my agents, when I checked
my CDR the caller-id shows my vendor's name and some shows as real customer
name.
When I call back again the real customer's number its reaching, the
answering machine owned by customer.
I have a confusion, or how
yes its not asterisk configuration, its phone feature and phone
configuration.
On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
So it is not at asterisk configuration?
Regards
Bilal
--
*From:* A J Stiles
If am not wrong even without doing any setting in asterisk side, if the
phone has Auto Answer it works.. !
Correct me if am wrong.
On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards asterisk@sedwards.comwrote:
Please don't top post.
On Wed, 17 Jul 2013, bilal ghayyad wrote:
So it is not
Hi,
Below link is the script which i found while surfing, this script basically
converts your voice file to flac format, where the file is reduced to 50%.
http://legroom.net/files/software/convtoflac.sh
The quality is really good, I tested. this...
In large production environment this script
If you want to store in external, why can't you have a NAS device and mount
to Asterisk server, let the mounted be a part in asterisk.conf, so that
voicemail will get recorded in external server...
Will it makes sense... !
Thanks.
On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe
, I suspect it could be a path issue.
Regards
On Fri, Jul 5, 2013 at 10:59 AM, Satish Barot satish4aster...@gmail.comwrote:
On Fri, Jul 5, 2013 at 1:45 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
exten = _4X.,1,Set(START_TIME=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten
-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
exten = _4X.,n,Dial(SIP/${EXTEN},30)
exten = _4X.,n,Hangup
Regards
On 4 Jul 2013 11:18, Satish Barot satish4aster...@gmail.com wrote:
On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I tried with hangup
Suddenly my asterisk restarted automatically and came up in seven seconds,
While checking core dump I see some message related to snmp.
No symbol table info available.
#5 0x7fc7e6249faa in agent_thread (arg=value optimized out) at
snmp/agent.c:206
__PRETTY_FUNCTION__ = agent_thread
#6
Ok thanks posting now
On 5 Jul 2013 03:09, Matthew Jordan mjor...@digium.com wrote:
On Thu, Jul 4, 2013 at 3:30 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Suddenly my asterisk restarted automatically and came up in seven seconds,
While checking core dump I see some message
*MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in
your dialplan.
Hope this helps.
--Satish Barot
Ahmedabad, India
On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi Satish,
I tried with sox, without any parameter, just
Am using Asterisk 11.2 in one location and 11.1 in another location.
when I trunk between two servers, the status is unreachable.
But with different server with 11.2 and 11.2 it works fine.
I tried both IAX and SIP.
the trunk in sip.conf what i have is,
[serverb]
type=friend
username=serverb
is there is no packet loss.. with mtr it is fine, tracepath is
fine, ping is fine... :(
On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Am using Asterisk 11.2 in one location and 11.1 in another location.
when I trunk between two servers, the status is unreachable
host=dynamic?
both servers are on 10.10.10.0 ? if no then check your deny permit seting.
On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Also tried one more scenario, particularly from one IP to other IP not
registering.
For example like 10.10.10.5
; hosts. This helps avoid the
configuration
; error of allowing your users to register
at
; the same address as a SIP provider.
On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote
/${EXTEN}) on side a.
make a call from a to b and one from b to and post cli log here or upload
anyware else.
On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
can't we use without register command both way as peer to peer?
On Wed, Jul 3, 2013 at 1:45 AM
} http://10.30.2.5/$%7BEXTEN%7D)
exten = _2XXX,n,Hangup
then you should handle the call when it arrive in any server
let me know if it work.
On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I tried creating two trunks with following,
*1st Location
By having different server, i made it work. I suspect some network issue...
On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad asghar...@gmail.com wrote:
make a call and post cli log
On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
still the peer shows
Am getting netsock error like this when using IAX2,
Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid =
4270)
== Using SIP RTP CoS mark 5
-- Executing [2001@Test:1] Dial(SIP/4090-0005,
SIP/2001@IAX2/IND-MAN,30)
in new stack
[Jun 23 06:31:36]
After changing my dialplan as suggested, there is no socket error, but
getting Busy/Congested, and the call is hanging up, let me check that
part...
Earlier my dialplan was,
;exten = _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30)
and I changed like this exten = _2XXX,1,Dial(${MANIAX}/${EXTEN},30)
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?
Regards,
Gopal.
--
...@digium.comwrote:
On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i
Hi Satish,
I tried with sox, without any parameter, just sox filename.wav to
filename.mp3, in linux shell prompt... the file is been converted...
Now If i want to run that command using dialplan,
MixMonitor(filename.wav,m)
Monitor_Exec(sox filename.wav filename.mp3)
Or to use System command?
I was go through'ing the following links for HA,
https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't
have file syncing.
https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster -
this one has file syncing with pacemaker
Any other HA applications available
Sometimes in huge call volume am facing this type of error,
[Jun 4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write:
Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun 4 08:43:04] WARNING[8285][C-79da]:
Asterisk 1.8 is stable
On 1 Jun 2013 16:40, luke devon luke_de...@yahoo.com wrote:
Hi
As I seen on the Asterisk web site , there is packages called ,
AsteriskLatest Version - 11.4.0
asterisk-11-current.tar.gzhttp://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
I am having a queue where included periodic announce like the below,
[test]
context = default
member = Agent/1001
member = Agent/1002
music = default
strategy = rrmemory
ringinuse = no
timeout = 15
retry = 1
maxlen = 0
joinempty = yes
leavewhenempty = no
periodic-announce =
It works.
Thanks
On 30 May 2013 19:39, Doug Lytle supp...@drdos.info wrote:
periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav
Try it without the .wav
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
Hi,
I am receiving DTMF without any reason after call establishment.
The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238]
So any resolution for this?
I suspect it could be related to RE INVITE
On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote:
i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.
On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N
Let me try with dtmfmode as auto...
On 28 May 2013 19:32, Asghar Mohammad asghar...@gmail.com wrote:
work around was block dtmf.
set wrong type of dtmf in incoming trunk.
On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
So any resolution for this?
I
With Asterisk 1.8 I got it working.
Regards
On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:
dtmfmode=auto
dtmfmode=info
or
dtmfmode
happens its keep on asking to enter digit If my DTMF didnt match. Do i
need to use any return function... ?
Actually my goal is to ask for 3 times and if not matched then return to
some other application.
Thanks in advance.
On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N
gopalakrishnan
= 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so
play badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
Thanks
On 25 May 2013 15:38, Gopalakrishnan N gopalakrishnan...@gmail.com
wrote:
Am using Read application to get the digit, since its
Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:
dtmfmode=auto
dtmfmode=info
or
dtmfmode=rfc2833
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
Hi,
Actually i would like to get the input from the user and he should not try
more than 3 times, he can try more than 3 times, if yes it will get routed
to the next priority and if not it goes to the loopback again from the
beginning.
And following is the one I created, I just want to know
I just want to make some increment... to 3 and yes go to the desired option
not to one more option.
On Thu, May 23, 2013 at 7:19 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Actually i would like to get the input from the user and he should not try
more than 3 times, he
488 not acceptable is due to codec error. Make sure you have right codec in
place between the end points.
On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker
m.grobec...@portunity.de wrote:
Hi,
Maybe you have not allowed T.38 as acceptable codec ;-)
You can try with allow=all in your
I have made the SIP bind port to 5070, and already I have one VoIP trunk
configured in my Asterisk 1.6.
Now the problem is after changing the bind port at some point of time, am
not able to dial in the DID number of the VoIP trunk!
Changing the bind port matters for this?
Regards.
--
@Marrie For one way audio as a debug strategy you can enable RTP debug and
see whether you have both way packets flow SENT and GOT.
Regards
On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer li...@jttech.se wrote:
2013-05-02 13:19, Marie Fischer skrev:
Hello everybody,
from time to time, we
Hi,
Has anybody worked on R2D Brazillian setup. I have configured R2 using
OpenR2 with Asterisk.
While doing some analysis I found R2D is already included in libopenr2.
Have anyone tested the same.
Regards,
Gopal.
--
_
--
and the
provider took the IP address from your PC and “locked out” the IP address
of your Asterisk server.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
*Sent:* Wednesday, September 26, 2012 7:51 AM
callwithus and configure your asterisk accordingly for
both accounts.
http://www.callwithus.com/configuration
BR
Sammy
On Thu, Sep 27, 2012 at 12:09 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I have registered in sip.conf and in my network i am not using any port
forwarding
Hi,
I was trying to register a VoIP trunk in Asterisk , where its keep on
sending Register message to the server, where I am not getting any response
from server.
But whereas if i register in Xlite softphone the account is getting
registered.
I suspect it could be network related issue, but
there.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
*Sent:* Wednesday, September 26, 2012 7:45 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] SIP Retransmitting REGISTER
, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
ahh... ! OK.. I though of this...
On Wed, Sep 26, 2012 at 6:24 PM, Danny Nicholas da...@debsinc.com wrote:
Another possibility – you registered from the softphone first and the
provider took the IP address from your PC and “locked out
, Patrick Lists wrote:
On 08/30/2012 09:45 AM, Gopalakrishnan N wrote:
Hi,
I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host,
I am not using any virtualbox, still i struck in loading the modules.
Please do not top post.
Install strace and then start asterisk
--
*From*: Gopalakrishnan N gopalakrishnan...@gmail.com
*Sent*: Tuesday, August 28, 2012 1:13 PM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
12.2
If I don't need
not load anything to start with so you can eliminate a rogue
module as the problem. Just change autoload=yes to autoload=no.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
*Sent:* Monday, August 27
like libpri is
biting you.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
*Sent:* Tuesday, August 28, 2012 11:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
: 186 modules
will be loaded.*
This is really tuff working with OpenSuse. I am clueless how to sort our
this.
Regards.
On Fri, Aug 24, 2012 at 3:55 AM, Hans Witvliet aster...@a-domani.nl wrote:
On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote:
Hi,
Again I stuck up with OpenSuse
, Gopalakrishnan N wrote:
This is really tuff working with OpenSuse. I am clueless how to sort our
this.
Maybe switch to a different distribution? I have used CentOS and RHEL for
years without any problems and as far as I know both debian and ubuntu
should work without problems too.
Regards,
Patrick
] *On Behalf Of *Gopalakrishnan N
*Sent:* Monday, August 27, 2012 8:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
12.2
** **
Hi Patrick,
** **
With other OS it works like charm. Only
.
Regards,
Gopal.
On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Ok Thanks Bryant, let me try with OpenSuse 12.1.
Regards.
On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote:
I have the current version of 8.x and 10.x on systems. I
Its really weird working with OpenSuse. I am not sure how others are using
with OpenSuse. Through Yast also I tried to install Asterisk package, it
didn't find.
Now I am clueless to work with OpenSuse.
Regards.
On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com
From the forum I understand OpenSuse 12.2 is pre-relase and better to use
OpenSuse 12.1. Lets check with OpenSuse 12.1.
Regards.
On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Its really weird working with OpenSuse. I am not sure how others are using
on all of our boxes are complied from source.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
--
*From*: Gopalakrishnan N gopalakrishnan...@gmail.com
*Sent*: Monday, August 20, 2012 10:11 AM
*To*: Bryant Zimmerman brya...@zktech.com
*Subject*: Re
Hi Patrick,
Thanks for your suggestion, even though I added my hostname in the
/etc/hosts, still the problem persists. Also I tried to install in OpenSuse
12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
hanging at modules while starting Asterisk.
Regards,
Gopal.
Hi,
I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
downloaded Asterisk 1.8 current version, after installing Asterisk, while
starting Asterisk it hangs at the stage of loading modules.conf, I checked
the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still
--
*From*: Gopalakrishnan N gopalakrishnan...@gmail.com
*Sent*: Monday, August 13, 2012 8:19 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Hi,
I am using
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