Dear Sir,
Can you confirm please if any version of asterisk does support ilbc 20ms
instead of 30 ms sample frequency?
Regards
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Hello Miguel molina,
I did what you asked, but still the voice is too bad
Regards
On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina wrote:
> El 05/08/10 14:50, Tim Nelson escribió:
>
> - "michel freiha" wrote:
> >
> > Dear Sir,
> >
> > I tri
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality
Regards
On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote:
> - "michel freiha" wrote:
> >
> > Dear All,
> >
> > i would like to ask please if someone tried to make a
Dear All,
i would like to ask please if someone tried to make a codec conversion from
ilbc to g729, because i did that but the voice quality was too bad and a lot
of disconnection..
Can i get your feedback regarding this issue please?
regards
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Hello,
The CPU load is great...It's 90% idle...even memory is great...60% Idle
Regards
On Sat, Jun 19, 2010 at 7:36 PM, Paul Belanger wrote:
> On Sat, Jun 19, 2010 at 5:21 AM, michel freiha wrote:
> > Waiting your reply
> >
> Reply: Do not cross-post to #asterisk-dev
Dear All,
I have installed 4 asterisks on the same Centos machine..>Each Asterisk has
its own installation folder and use its own libraries...Everything looks
great and all asterisks are doing their jobs correctly except one thing...I
faced a voice quality issue...On a specific time, and after the
Hi all,
I would like to ask please how to configure asterisk in order to unforce
rtp traffic to pass through it and send them to a separate RTp proxy?
Regards
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valid handler
regards
On Mon, Sep 28, 2009 at 5:52 PM, Doug Lytle wrote:
> michel freiha wrote:
> > Hi all,
> >
> > I need a goto If statement syntax that check if a variable is not
> > null then go to dialplan 1 else go to dialplan2
> >
>
> exten =>
Hi all,
I need a goto If statement syntax that check if a variable is not null then
go to dialplan 1 else go to dialplan2
Regards
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Dear All,
I have a perl script running on my asterisk server...This script is running
for all incoming calls...It checks if a user is registered on openSIPS
server...Else it return a busy tone...
I would like to ask you please about the syntax to call a dialplan defined
on extensions.conf for a s
Dear Sir,
I'm trying to install asterisk 1.6.1.1 on solaris 10...At the end of gmake I
got the below error
creating config.h
In file included from sig.h:47,
from el.h:107,
from common.c:51,
from editline.c:4:
/usr/include/signal.h:77: error: syn
ask ff00 broadcast 192.168.0.255
ether 0:3:ba:f2:d2:ea
Yes I have a NIC, Up and running and I can SSH the server from that NIC
Regards
On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro
wrote:
>
>
> On Fri, Jul 17, 2009 at 2:08 AM, michel freiha wrote:
>
>> Hi all,
>>
&g
Hi all,
Can you please let me know what the below issue mean when trying to start
asterisk and how I can fix it?
pbx_dundi.c: No ethernet interface found for seeding global EID You will
have to set it manually.
regards
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Hi all
I'm trying to install asteris 1.4.22.1 on Solaris 10...the server is V120
SUN spark...During compilation (gmake) I got the following error
/vis.c -o np/vis.o_a
np/vis.c: In function `svis':
np/vis.c:205: error: `u_int32_t' undeclared (first use in this function)
np/vis.c:205: error: (Each
n asterisk directly the authentication is
successfully done by asterisk
regards
On Mon, Jun 29, 2009 at 6:06 PM, Alex Balashov wrote:
> It does this by default unless you have allowguest set to yes, and/or
> any insecure parameter options on any individual peers.
>
> --
&
Hi all,
i would like to ask please about how to force asterisk to ask for
authentication when receiving an INVITE packet from any device?
Regards
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Hi all,
I'm using asterisk in Realtime mode and all my dialplans are defined in
extensions table...My problem is the following...
If I have for a specific context an extension with value _X. and another
entry for the same context with extension 123456, if an incoming call from
the same context com
ook like?
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, May 14, 2009 14:13
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-use
Hi all,
I'm using asterisk in realtime...I have a specific scenario to jump from
context to another context...The call will come from a gateway registered
under Test context and this call should be sent to the On-net extension as
Listed in the paste bin below:
http://pastebin.com/d50b2ba42
The i
Hi all,
Does asterisk support the following scenario? I need when a customer who own
an endpoint registered on asterisk open the line, the asterisk will run a
specific AGI script inside the endpoint context?
Regards
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.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Wednesday, April 08, 2009 4:28 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Perl AGI
>
>
>
> Hi all,
>
> I have the belo
Hi all,
I have the below peace of my AGI script...the problem here is that I cannot
fetch the extension value to inside the script and assign it to another
variable...I highlighted it in red
#!/usr/bin/perl
#use DBD::mysql;
use DBI;
use DBD::mysql;
use Asterisk::AGI;
Dear All,
I'm trying to send FAX to an endpoint Behind NAT...The scenario i the
following:
PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT..
The FAX is failed and I got the following error log on asterisk:
Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite:
Hi all,
I mentioned in asterisk.conf there is a property "maxcalls"...I know that
this is the max number of concurrent calls but i need to know please if this
entry is commented out, what is the default number of MAX concurrent calls
supported by asterisk?
Regards
_
Hi all,
I'm getting the following error when receiving a FAX on Asterisk...
ERROR[11320]: chan_sip.c:12441 handle_response_invite: Got error on T.38
re-invite. Bad configuration. Peer needs to have T.38 disabled.
Any suggestions?
Regards
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Hi all,
Does anyone knows how to add a new variable to the predefined variables
sent by asterisk to AGI script?
regards
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Dear All,
I have the following warnings on my log file:
[Mar 3 15:26:23] NOTICE[22639] chan_local.c: No such extension/context
0.0@default creating local channel
[Mar 3 15:26:23] WARNING[22639] app_dial.c: Unable to create channel of
type 'LOCAL' (cause 0 - Unknown)
[Mar 3 15:26:24] WARNING
Hi all,
I'm using asterisk in real time mode...My extensions.conf table contains:
[default]
switch => Realtime/defa...@extensions
I have added the following to extensions.conf table;
context:micho
exten: _X.
priority: 1
app:Dial
appdata: SIP/00xxx...@pstn GAteway
Asterisk server is connected s
Dear Sir,
The issue has been solved
rtcachefriends=no
\and everything will work
Thanks
On Mon, Mar 2, 2009 at 10:31 PM, michel freiha wrote:
> Hi all,
>
> I'm using asterisk in real time mode...All extensions are defined in table
> sip_buddies...Everything looks fine and as
Hi all,
I'm using asterisk in real time mode...All extensions are defined in table
sip_buddies...Everything looks fine and asterisk is reading extensions info
from the sip_buddies table...The problem occurs as soon as any information
on an extension is changed from sip_buddies table...Which mean,
Dear Keven,
I have just post a new email with the same body due to a member advice
Regards
On Sun, Mar 1, 2009 at 8:21 PM, Kevin P. Fleming wrote:
> michel freiha wrote:
> > Dear All,
> > I have created an inbound context in sip.conf that forward incoming call
> > to
Dear All,
I have created an inbound context in sip.conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl = yes
under General context...The Asterisk negotiate the SIP session with OpenSIPS
without adding voice codec to INVITE packet...It just contai
face such issue
Regards
On Sat, Feb 28, 2009 at 7:21 PM, michel freiha wrote:
> Hi all
> I'm using asterisk for making PSTN calls from extensions registered on
> OpenSIPS...In peak hours ,number of calls Increase dramatically to a non
> logic number..When checking the calls
Hi all
I'm using asterisk for making PSTN calls from extensions registered on
OpenSIPS...In peak hours ,number of calls Increase dramatically to a non
logic number..When checking the calls using asterisk CLI I saw a lot of
calls in ringing status and after 300s(rtphold timeout), asterisk releas
Dear David,
Please find on http://pastebin.com/m69b8559d my sip.conf file
Thanks a lot
On Fri, Feb 27, 2009 at 1:05 PM, David fire wrote:
> paste your sip.conf.
> David
>
> 2009/2/26 michel freiha
>
>> Dear All,
>> I have created an inbound context in SIP .conf
Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38 protocol...Wh
Dera All,
I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I
Dear All,
I would really need to thank you all for the great help that I got here from
all of you and specially Mr. Yawar hadi for his great assist and
professionalism
Thanks
On Mon, Feb 23, 2009 at 6:48 PM, Steve Edwards wrote:
> On Mon, 23 Feb 2009, Yawar Hadi wrote:
>
> > so if u want to rea
to read extension then supplu variable name like
> $myno=$AGI->get_variable('EXTEN');
> hope u get it
>
>
>
> On Mon, Feb 23, 2009 at 1:14 PM, michel freiha wrote:
>
>> Dear Sir,
>>
>> Kindly note that the problem is on command $AGI->get_variable(
t;
>
> On Mon, Feb 23, 2009 at 12:23 PM, Yawar Hadi wrote:
>
>> How ?
>>
>>
>> On Mon, Feb 23, 2009 at 12:09 PM, Steve Edwards > sedwards.com> wrote:
>>
>>> On Mon, 23 Feb 2009, Yawar Hadi wrote:
>>>
>>> > dear s
>>
>> Luis Morales.
>>
>> On Fri, Feb 20, 2009 at 7:24 PM, michel freiha wrote:
>> > Dear Sir,
>> >
>> > I need the followingA customer will dial a specific number like
>> 112,this
>> > will fire a php AGI script...The AGI script wi
Hi all,
I took my decision to use Asterisk server for handling my VOIP calls...My
next step is to choose the best hardware that I should use i order to have
the best performance...Here I faced 2 choices for my hardware (CPU)...
1- Using Intel CPU or AMD
2- Use 32 or 64 bits
Can you help me please
Hi all,
Here I a have a question and hope that someone give me the right answer...Is
it better to use Intel CPU inside the hardware where I need to install
Asterisk or AMD? It's better to use 32 bits or 64 bit and what is the
difference between both of them?
Thanks a lot
On Thu, Feb 19, 2009 at
hi all,
I have the script but i have a small issue and need some help..can you help plz?
On 2/21/09, Geoff Lane wrote:
> Hi All,
>
> Asterisk 1.4.12 on CentOS 5
>
> I'm trying to increment an AstDB key with the length of the last
> outgoing call. Here's what I've got for "01" UK geographical numb
nly written as user instead of /user on first line let
>>>> be go through the problem of not reading the vairable and reply back to you
>>>> soon.
>>>> wait and dont lose your interest .this is the way to learn some thing
>>>> new .wait let me to c
FROM test where extension = ''
regards
On Sat, Feb 21, 2009 at 12:06 AM, michel freiha wrote:
> Dear All,
>
> i found the issue...it was just the #!/user/bin/perl instead of
> #!user/bin/perl
>
> Thanks a lot for all help
>
> Regards
>
> On Fri, Feb 20,
Dear All,
i found the issue...it was just the #!/user/bin/perl instead of
#!user/bin/perl
Thanks a lot for all help
Regards
On Fri, Feb 20, 2009 at 11:58 PM, michel freiha wrote:
> Dear Kai,
>
> If I change directory to /usr/bin and type the following command I got:
>
> [r
Dear Kai,
If I change directory to /usr/bin and type the following command I got:
[r...@switch1 bin]# ./perl /var/lib/asterisk/agi-bin/dial.pl
VERBOSE "AGI Environment Dump:" 3
GET VARIABLE extension
VERBOSE "my dialed no is :"
VERBOSE "Query is:SELECT dst_nb FROM test where extension = ''"
?
Thanks
On Fri, Feb 20, 2009 at 10:23 PM, Steve Edwards
wrote:
> On Fri, 20 Feb 2009, michel freiha wrote:
>
> > The file dial.pl is executable and own by root as you can see below
> >
> > ls -lrt
> > -rwxrwxrwx 1 root root 1865 Feb 20 21:18 dial.pl
>
> Usin
Fri, Feb 20, 2009 at 10:15 PM, Steve Edwards
wrote:
> On Fri, 20 Feb 2009, michel freiha wrote:
>
> > I'm facing the below problem:
> > == dial.pl|112: Failed to execute '/var/lib/asterisk/agi-bin/dial.pl':
> No
> > such file or directory
> >
>
Sorry, it's
use Asterisk::AGI;
On Fri, Feb 20, 2009 at 10:13 PM, michel freiha wrote:
> Kindly note that at line 6 I have:
>
> se Asterisk::AGI;
>
> Regards
>
> On Fri, Feb 20, 2009 at 9:44 PM, Danny Nicholas wrote:
>
>> Dial.pl is
>>
>>1
t; *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Friday, February 20, 2009 1:41 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] AGI script
oun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Friday, February 20, 2009 1:41 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] AGI script
>
>
>
> Dear Sir,
>
> I'm facing the below problem:
>
&g
to
> which u r sending ur traffic.if u want to call sip sip of from one soft
> phone to second soft phone then remoce /terminator ok... like
> (SIP/${DestNo})
>
> mail be back if any problem.
>
>
> On Fri, Feb 20, 2009 at 1:07 PM, michel freiha wrote:
>
>> Dear Yawar
Dear Sir,
Kindly note that I'm getting the following error when trying to register on
asterisk server:
[Feb 20 17:12:32] WARNING[2865]: res_config_mysql.c:358 update_mysql: MySQL
RealTime: Failed to query database. Check debug for more info.
I can connect to the database using the username and p
>> On Fri, 20 Feb 2009, michel freiha wrote:
>>
>> > I need the followingA customer will dial a specific number like
>> 112,this
>> > will fire a php AGI script...The AGI script will connect to the database
>> and
>> > see if this number (112)
registered in that table
and match the 112 then asterisk will dial out the number fetched
Please let me know if you need any other information
Regards
On Fri, Feb 20, 2009 at 1:38 AM, Luis Morales wrote:
> Sure,
>
> what did you need exactly ?
>
>
> On Thu, Feb 19, 2009 at 6:57 PM, mi
Dear All,
I would like to ask please if someone has a AGI script that select a value
from a database and dial this value as a destination number
Regards
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elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
> asterisk->support books section Asterisk: The Future of
> Telephony<http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf>
> is greate.
> David
>
> 2009/
Dear Sir,
Can someone help me please to find a free ebook talking about AGI scripting
through asterisk?
Regards
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Dear Helm,
Kindly confirm why you do not recommend the VMs solution and if you had bad
experience for it and what did you get?
Regards
On Tue, Feb 17, 2009 at 9:24 PM, Wilton Helm wrote:
> >You may be able to split up some of the servers into multiple VMs -- maybe
> five >servers with five VMs
erisk purely as a transit
> element for billing. Just because a2billing is available does not mean
> you should. Far more scalable solutions are easily available.
>
> --
> Sent from mobile device
>
> On Feb 17, 2009, at 10:19 AM, michel freiha wrote:
>
> > Hi all,
&g
Hi all,
I'm planning to build a VOIP solution for handling SIP calls coming from
endpoints registered on a specific SIP proxy...I made some research
regarding network architecture and found out that the best solution is to
use OpenSips as SIP proxy for registration and local calls between
register
Hi all,
I'm using Asterisk in real-time mode...i need to limit the number of
outgoing concurrent call per extension...Wich mean limit the number of
concurrent outgoing calls to 2 at a time...I added a call-limit field to
sip_buddies table and put it as 2 for an extension...I tried to make 3
concur
Thanks Johansson,
Everything is OK now and you were right
Regards
On Thu, Feb 12, 2009 at 12:30 PM, Johansson Olle E wrote:
>
> 11 feb 2009 kl. 10.43 skrev michel freiha:
>
> > Hi all,
> > I need to register asterisk on an OpenSIPS SIP Proxy...The
> > Registrati
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you c
263.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, January 29, 2009 2:10 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [aster
Then just do a make && make install on asterisk again.
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, January 29, 2009 1:35 PM
>
*From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, January 29, 2009 1:35 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] FAX
>
>
>
I'm getting now the below notice:
rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha wrote:
> Do you mean call limit on the extension or on the outgoing gateway? Kindly
> note that my outbound dialpeer has m
ll-limit is important because 1 call can actually
> be 2-N hops.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, January 29, 200
; asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Thursday, January 29, 2009 12:27 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] FAX
>
>
>
> Dear Sir,
>
>
>
> When trying
] chan_sip.c: Hanging up call
3058f601-47504...@14.14.14.49 - no reply to our critical packet (see
doc/sip-retransmit.txt).
Regards
On Thu, Jan 29, 2009 at 12:04 AM, michel freiha wrote:
> Dear Danny,
>
> Thanks a lot for the help...I'll try and let you know
>
> Regards
>
&
> may slow down the process or cause undesirable results by using/accounting
> for unneeded or unwanted codecs.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel
gt;
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
> *Sent:* Wednesday, January 28, 2009 3:09 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-use
try
> adding gsm or just comment out the disallow and the 2 allows. (your
> recipient is using a codec that isn't ulaw or alaw).
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium
um.com] *On Behalf Of *michel freiha
> *Sent:* Wednesday, January 28, 2009 9:30 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] FAX
>
>
>
> Hi all,
>
> When trying to send a FAX I got the following error:
>
> Executi
Hi all,
When trying to send a FAX I got the following error:
Executing [003228949...@micho:1] Dial("SIP/028949469-08466918", "SIP/
003228949...@80.169.210.181|60") in new stack
[Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format
found to offer. Cancelling call to 003228949
Dear All,
I'm trying to send Fax using T.38 protocol but the FAX is not going
through..I'm getting the following error om /var/log/messages
[Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from
SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256)
[Jan 27 16:46:40] WARNING
Dear Sir,
I would like to ask please about how I can force asterisk to send all G726
codecs without translation...
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723- ---- -- -- --
--
gsm- -22
Dear Jeff,
Please find attached the autoprovisionning file of the PAP2T...Kindly let me
know if you need to know how to use it
Regards
On Wed, Jan 21, 2009 at 1:04 AM, Jeff LaCoursiere wrote:
>
> Anyone have an example XML file for the PAP2T?
>
> Cheers,
>
> j
>
> _
c/cpuinfo
>
> 2009/1/19 michel freiha
>
>> Dear All,
>>
>> I have the following CPU info on my asterisk server:
>>
>> Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43
>> EST 2008 i686 i686 i386 GNU/Linux
>>
>> I need t
Dear All,
I have the following CPU info on my asterisk server:
Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
2008 i686 i686 i386 GNU/Linux
I need to install G729 on the asterisk server just to pass through and not
for encoding...Which G729 package do you advice me to
Dear Sir,
I used several other Softphones like Skype and they are facing the same
problem...It seems that the issue is global du to an undersea cable cut
Regards
On Mon, Dec 22, 2008 at 9:07 PM, michel freiha wrote:
> Hi all,
> Sometimes when making a PC to PSTN call through asterisk, I
Hi all,
Sometimes when making a PC to PSTN call through asterisk, I got no audio in
both sides...tracing by wireshark, I can find that RTP packets are hitting
my PC but no audio...Can someone guess what could be that issue?
Maybe it's a latency issue?
Regards
_
fromCustomer_IP:15934 (type 18, seq 038497, ts
3706128855, len 20)
Regards
On Thu, Dec 18, 2008 at 10:58 AM, michel freiha wrote:
> Dear All,
>
> I would like to ask please if there is any buffer for RTP packets sent
> through asterisk and if there is a way to change the value
Dear All,
I would like to ask please if there is any buffer for RTP packets sent
through asterisk and if there is a way to change the value in order to
increase it
Regards
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asteri
ch packet has a timestamp at the
> beginning measured in ten thousandths (I think?) of a second. You should
> be able to see the RTP packet arrive and then leave again... just subtract
> the timestamps for your added latency.
>
> Cheers,
>
> j
>
> On Mon, 15 Dec 2008, michel frei
Dear Sir,
I would like to ask please where I can find the results for RTCP debug
on...It's saved on a file or it appears in the CDRs?
Regards
On Tue, Dec 16, 2008 at 12:06 AM, Mark Michelson wrote:
> michel freiha wrote:
> > Dear Sir,
> >
> > What I'm interested
ds out packets with continually
> increasing TTLs and the router that drops the packet will send back a
> notification, so you can "trace" each hop...
>
> What is it you are trying to do or measure?
>
> j
>
> On Mon, 15 Dec 2008, michel freiha wrote:
>
> > Dear
bytes in the packet.
>
> j
>
> On Mon, 15 Dec 2008, michel freiha wrote:
>
> > *Dear All,
> > I run the below tcp dump on my asterisk server
> >
> > tcpdump -i eth0 -n -s0 -v udp port 5060
> >
> > I got the following result
> >
> > 20:29:
*Dear All,
I run the below tcp dump on my asterisk server
tcpdump -i eth0 -n -s0 -v udp port 5060
I got the following result
20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17,
length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345
What i need to know please
ack tone all the
time...Is that possible?
Regards
On Sat, Dec 13, 2008 at 10:40 AM, Philipp Kempgen wrote:
> Andrew Joakimsen schrieb:
> > On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling
> wrote:
> >> Philipp Kempgen wrote:
> >>> michel freiha schrieb
Hi all,
I would like to ask please if there is a way to play a ring back tone from
asterisk when the customer try to make a call...I already added the ringing
function to the context in extensions .conf and it work perfectly...But the
issue that the asterisk server is stoping playing back his own
Dear All,
I would like to ask please if there is a way to reduce latency on asterisk
or to check what is causing this latency
Regards
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Dear Sir,
My Asterisk server is sending periodically the below SIP packets
Retransmitting #4 (NAT) to 68.62.168.138:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK37f337ed;rport
From: "asterisk" ;tag=as078bf319
To:
Contact:
Call-ID: [EMAIL PROTE
Dear Sir,
I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk
-rv I can see a lot of messages about RTCP too short...
-- Remote UNIX connection disconnected
[Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too
short
[Nov 28 13:33:00] WARNING[1980
Hi all,
Kindly note that I got the below message when sending DTMF in RFC2833
through asterisk PBX...The DTMF is not going through
RTCP Read too short
I'm using G729 codec and asteriks Asterisk 1.4.21.2
Regards
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Dear All,
Kindly let me know please where I can fix the payload of DTMF to 101...I'm
using RFC2833
Regards
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Dear Sir,
I need to configure my Voice Mail on asterisk...I made the following
configuration:
*
extensions.conf:*
exten => _999.,1,VoiceMail(${EXTEN})
exten => _999.,2,HangUp()
If the customer dial 9991234 then a prompt message should ask him to enter
his voice message and this what is not happe
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