Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this not true? is there any way to do what I want? Thanks! --ian
/SNIP/ Subject: [Asterisk-Users] Codec Selection I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this not true? is there any way to do what I want? /SNIP/ My Guess is that you need two licenses in order to call from both ends, where both end point are devoid of g729 and you need transcoding for both channels. Seshu -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
Hi All, I have one Carrier which is supporting only G.723.1, how i can put in my extentions.conf to send calls to this GW using G.723.1, because for Clients i can specify the codec from sip.conf but i am little confiuse how i can give specific codec for carriers. your ideas will be appriciated. -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hi, I guess what you mean by a Carrier as Trunk. If you have an SIP Trunk i feel the preference list will do the needful. disallow=all allow=g723 Dan On 05/02/06, Abdul Lateef <abdul_zu@yahoo.com> wrote:> > Hi All, > > I have one Carrier which is supporting only G.723.1, > how i can put in my extentions.conf to send calls to > this GW using G.723.1, because for Clients i can > specify the codec from sip.conf but i am little > confiuse how i can give specific codec for carriers. > > your ideas will be appriciated. > > > > -------- > Yours, > Abdul Lateef > Computer Programmer > HATIF COM > Mob: +974 - 5405022 > ICQ: 276994704 > MSN: abdulzu@hotmail.com > GoogleTalk: lateef.np@gmail.com > YM!: abdul_zu > Doha Qatar > http://www.hatif.com > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060205/c925eaa4/attachment.htm
Hi, We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. Thank you in advance! Phil
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|30|tTL(" . ($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
--------
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: abdulzu@hotmail.com
GoogleTalk: lateef.np@gmail.com
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
-----------------------------------
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to
transcode on asterisk.
If client has that codec,asterisk will do pass-thru
and it will work.
Olivier
-----Message d'origine-----
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoy? : dimanche 5 f?vrier 2006 20:00
? : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI->get_variable("DIALSTATUS");
if ($discr == "CONGESTION" || $discr == "NOANSWER" ||
$discr == "CHANUNAVAIL")
{
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|30|tTL(" . ($crdeit*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
$discr = "";
}
Any idea?
--------
--------
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: abdulzu@hotmail.com
GoogleTalk: lateef.np@gmail.com
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
> Hi, > > We have 10 people on our network and each person will have a SIP phone > connected to our Asterisk server. All phones, Asterisk, other servers and > users workstations will be using the same network. The question is: would > I need a QOS device to give SIP traffic a chance? Our internal network is > 100M. We will have a ISDN30 for outgoing calls. No calls will be made > over the internet. >As long as the current infrastructure is decent, you should be fine without a separate voice switch. PaulH
On 12:50, Sun 05 Feb 06, Abdul Lateef wrote:> > Hi, > > Is there any special configuration for transcoding on > asterisk? Or Asterisk will do it automatically?If the codecs from both ends are known to asterisk, * will do it automagically for you :) -- Michiel van Baak http://michiel.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"
What will be the g729 and g723 codec capacity from Intel IPP liberary without License? Because still i am developing all billing and other application for asterisk so first i want to use these codecs for test once all our system become stable i will buy the license. S0 please let me know how many cuncurent calls can be handel using Intel IPP? -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hi Abdul You will need to download and install the Intel API which is then used to compile the patched G723 codec. Hope this helps. Kind Regards Garth Abdul Lateef wrote:> Hi All, > > I have one Carrier which is supporting only G.723.1, > how i can put in my extentions.conf to send calls to > this GW using G.723.1, because for Clients i can > specify the codec from sip.conf but i am little > confiuse how i can give specific codec for carriers. > > your ideas will be appriciated. > > > > -------- > Yours, > Abdul Lateef > Computer Programmer > HATIF COM > Mob: +974 - 5405022 > ICQ: 276994704 > MSN: abdulzu@hotmail.com > GoogleTalk: lateef.np@gmail.com > YM!: abdul_zu > Doha Qatar > http://www.hatif.com > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Garth van Sittert BSc (Physics & Computer Science) ----------------- Mobile: +27 (0)83 791 6662 Email: garth@bitco.co.za Phone: 08600 BITCO Web: www.bitco.co.za