RES: [Asterisk-Users] Commercial g723.1 license for asterisk
Why not? you only need to pay US$30.500 plus US$1.78 per port to use. Oh, you also need to pay US$199.00 to license IPP library. http://www.dspg.com/technology/LicensePricing.html http://www.intel.com/software/products/ipp/pricelist.htm -Mensagem original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de E. VersaevelEnviada em: terça-feira, 23 de novembro de 2004 10:09Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'Assunto: RE: [Asterisk-Users] Commercial g723.1 license for asterisk Which is not for commercial use Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Vinicius VianaVerzonden: dinsdag 23 november 2004 14:57Aan: Asterisk Users Mailing List - Non-Commercial DiscussionOnderwerp: RES: [Asterisk-Users] Commercial g723.1 license for asterisk I never used, but there are one "open" beta codec g723.1 from the makers of the "open" g729 codec that uses Intel IPP library. http://www.readytechnology.co.uk/open/g723.1/ Regards, Vinicius -Mensagem original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de kido noagbodjiEnviada em: terça-feira, 23 de novembro de 2004 09:34Para: Asterisk Users Mailing List - Non-Commercial DiscussionAssunto: [Asterisk-Users] Commercial g723.1 license for asterisk Hi all, Is there any commercial g723 license for asterisk? Where can it be purchased? Has somebody used it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Commercial g723.1 license for asterisk
I never used, but there are one "open" beta codec g723.1 from the makers of the "open" g729 codec that uses Intel IPP library. http://www.readytechnology.co.uk/open/g723.1/ Regards, Vinicius -Mensagem original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de kido noagbodjiEnviada em: terça-feira, 23 de novembro de 2004 09:34Para: Asterisk Users Mailing List - Non-Commercial DiscussionAssunto: [Asterisk-Users] Commercial g723.1 license for asterisk Hi all, Is there any commercial g723 license for asterisk? Where can it be purchased? Has somebody used it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: RES: [Asterisk-Users] 403 Forbidden
The call end reason "EndedByQ931Cause" is used by the OpenH323 stack when it doesn't know the real cause. Try to see if the codecs in the gateway are compatible with the codecs in asterisk. What are the codecs you are using in SIP Phones, in Asterisk and in the gateway? Regards, Vinicius -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de jesus Enviada em: quinta-feira, 11 de março de 2004 11:37 Para: [EMAIL PROTECTED]; Vinicius Viana Assunto: Re: RES: [Asterisk-Users] 403 Forbidden Hi, thanks a lot for your answer. When I call from SIP phone to analogic found I get this log file: (I only show, when there's the disconnection) 46:01.165 H245:816f650 H245Received capability set, is accepted 46:01.165 H245:816f650 H245TerminalCapabilitySet already in progress: outSeq=1 46:01.165 H245:816f650 H245Sending PDU: response terminalCapabilitySetAck 46:01.166 H245:816f650 H323 InternalEstablishedConnectionCheck: connectionState=Await ingSignalConnect fastStartState=FastStartDisabled 46:01.167 H225 Caller:8141218 H225Set protocol version to 4 46:01.167 H225 Caller:8141218 H323Clearing connection ip$localhost/7705 reason=EndedByQ931C ause 46:01.167 H225 Caller:8141218 H323Call end reason for ip$localhost/7705 set to EndedByQ931C ause 46:01.167 H225 Caller:8141218 H225Sending release complete PDU: callRef=7705 46:01.170 H225 Caller:8141218 H245Sending PDU: command endSessionCommand 46:01.170 H225 Caller:8141218 H225Sending PDU: releaseComplete 46:01.171 H323 Cleaner H323Cleaning up connections I suppose, from what you have told me in your mail, that the problem is in my gateway so, have you any idea what can be the exact problem and how to solve it? Thanks a lot for you answer. Best Regards, Mireia Quoting Vinicius Viana <[EMAIL PROTECTED]>: > I believe your gatekeeper or your gateway is refusing the call. This can be > a authorization problem in the gatekeeper or codec problem in the gateway. > > You need to see where your call is failing. Try to do the following: > > 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to > your configuration: > wrapLibTraceLevel=3 > libTraceLevel=3 > libTraceFile=/var/log/asterisk/oh323.log > > 2 - Make a call from your SIP Phone to your PBX > > 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is > failing in the Admission Request or in the Setup message. > > 4 - If it fails in the Admission Request (you will see a Admission Reject > into the log) the problem is in the configuration of your gatekeeper. > 5 - If it fails in the Setup message (you will see a Release Complete into > the log) the problem is in the configuration of your gateway > > Other thing you can see is if your asterisk box is registered with your > gatekeeper. > > With the information you supplied this is what I remember you can check to > see what is wrong. > > Regards, > > Vinicius > > -Mensagem original- > De: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de > jesus > Enviada em: quarta-feira, 10 de março de 2004 16:46 > Para: [EMAIL PROTECTED]; Martin Mielke > Cc: [EMAIL PROTECTED] > Assunto: Re: [Asterisk-Users] 403 Forbidden > > > Hi, > > Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway > and > calls between SIP clients (phone and soft clients) are working all right. > The > only problem I have, is like I have said in my mail is between sip phones > and > PBX. > > Best Regards, > > Mireia > > PS: Someone have other ideas? > > > Quoting Martin Mielke <[EMAIL PROTECTED]>: > > > Hi Mieria, > > > > Mireia Munoz de jesus wrote: > > > > >Hi! > > > > > >When I try to call from a SIP phone to a PBX phone I get this error: > > > > > >chan_oh323.c [1004] Couldn`t call 483377839 > > > > > >and if I get the messages from SIP debug, I have a 403 message. The > > >configuration of my system is: > > > > > >SIP Phone ASterisk Gatekeeper - Gateway - PBX - > Phone > > > > > >Have someone any idea of what is going on?. It will be very nice if > someone > > >helps... it`s been more than a week that I can`t solve this problem. > > > > > >Best Regards, > > > > > >Mireia > > > > > > > Could it be that you are using a *SIP* phone? Although you can add > > H.323 to Asteriskm, SIP and H.323 are different protocols... > > > > > > HTH,
RES: [Asterisk-Users] 403 Forbidden
I believe your gatekeeper or your gateway is refusing the call. This can be a authorization problem in the gatekeeper or codec problem in the gateway. You need to see where your call is failing. Try to do the following: 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to your configuration: wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/var/log/asterisk/oh323.log 2 - Make a call from your SIP Phone to your PBX 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is failing in the Admission Request or in the Setup message. 4 - If it fails in the Admission Request (you will see a Admission Reject into the log) the problem is in the configuration of your gatekeeper. 5 - If it fails in the Setup message (you will see a Release Complete into the log) the problem is in the configuration of your gateway Other thing you can see is if your asterisk box is registered with your gatekeeper. With the information you supplied this is what I remember you can check to see what is wrong. Regards, Vinicius -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de jesus Enviada em: quarta-feira, 10 de março de 2004 16:46 Para: [EMAIL PROTECTED]; Martin Mielke Cc: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] 403 Forbidden Hi, Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway and calls between SIP clients (phone and soft clients) are working all right. The only problem I have, is like I have said in my mail is between sip phones and PBX. Best Regards, Mireia PS: Someone have other ideas? Quoting Martin Mielke <[EMAIL PROTECTED]>: > Hi Mieria, > > Mireia Munoz de jesus wrote: > > >Hi! > > > >When I try to call from a SIP phone to a PBX phone I get this error: > > > >chan_oh323.c [1004] Couldn`t call 483377839 > > > >and if I get the messages from SIP debug, I have a 403 message. The > >configuration of my system is: > > > >SIP Phone ASterisk Gatekeeper - Gateway - PBX - Phone > > > >Have someone any idea of what is going on?. It will be very nice if someone > >helps... it`s been more than a week that I can`t solve this problem. > > > >Best Regards, > > > >Mireia > > > > Could it be that you are using a *SIP* phone? Although you can add > H.323 to Asteriskm, SIP and H.323 are different protocols... > > > HTH, > > Martin > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] TE410P cards E1 ports not working!
You are informing for Asterisk that your E1 channels will use FSX Loop Start signalling and the boards are expecting ISDN PRI signalling. Try to put a PRI_CPE signalling in your configuration. Something like that: . . . group=2 switchtype=euroisdn signalling=PRI_CPE pridialplan=international context=incoming channel=49-63,65-79,0-94,96-110 . . . Regards, Vinicius -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Augustine Olaifa Enviada em: quarta-feira, 10 de março de 2004 06:07 Para: [EMAIL PROTECTED] Assunto: [Asterisk-Users] TE410P cards E1 ports not working! we have just got a TE410P digium card. and i am tryng to install it. i have installed it and loaded the driver accordingly, but i can't get the E1 ports to work. This s the scenerio, i want to configure 2 T1s(40fxs and 8fxo) and 2 E1s,but when i run asterisk get : ERROR [8192] -chan_zap.c:5336 mkintf signalling request is FXs loopstart but lne s in RI signalling signalling ERROR[8192] -chan_zap.c:7397 setup_zap: unable to register channel 49-63 my zaptel.conf is: span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,1,0,ccs,hdb3 span=4,2,0,ccs,hdb3 fxols=1-24 fxsls=25-32 fxols=33-48 bchan=49-63 dchan=64 bchan=65-79 bchan=80-94 dchan=95 bchan=96-110 zapata.conf [channels] language=en context=incoming cntext=internal group=1 sgnalling=fxo_ls context=internal channel=1-24 channel=33-48 signallng=fxs_ls context=incoming channel=25-32 group=2 switchtype=euroisdn pridialplan=international context=incoming channel=49-63,65-79,0-94,96-110 i do not know what i am doing wrong? have also played aroound with the timing making it "0" instaed of "1" and "2".it still did not work. regards -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] SIP Conference Bridge?
Asterisk needs a timing source to do conference. If you don't have a Zaptel interface installed but have a uhci-usb interface you can use the ztdummy to generate the timing. To use it you need to go to the zaptel driver directory and modify the Makefile removing the double minus (--) in front of ztdummy.c After this do a make clean and a make install and modprobe ztdummy Other thing, in order to ztdummy works you need to have the uhci-usb installed as a module and not in the kernel. Regards, Vinicius -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Todd R. Stroup Enviada em: segunda-feira, 8 de março de 2004 17:04 Para: [EMAIL PROTECTED] Assunto: [Asterisk-Users] SIP Conference Bridge? Can Asterisk act as a SIP conference bridge? Looking through the source I notice that it's required to have a Zaptel interface installed. Why is this a requirement? Can you not mesh the VoIP streams together? Thanks, T..S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
