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From: Meftah T. <tay...@gm...> - 2010-05-30 12:51:26
|
hi, if you don't want to migrate to cvs, try to build a mirore for SVN/Git in sourceforge i think sourceforge can do it cvs is very complicated Le 30/05/2010 02:14, Joegen E. Baclor a écrit : > Hi Meftah, > > The main reason is CVS does the job and there are current build > systems (Solegy's) is wired to sync the SRC via CVS. Any particular > reason why we need to migrate to SVN/Git? > > Joegen > > > On Monday, 31 May, 2010 03:20 AM, Meftah Tayeb wrote: >> hi hfolk, >> why we don't migrate opensipstack to svn or git? >> >> ------------------------------------------------------------------------------ >> >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> > |
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From: Joegen E. B. <jo...@op...> - 2010-05-30 00:24:10
|
Hi Meftah, The main reason is CVS does the job and there are current build systems (Solegy's) is wired to sync the SRC via CVS. Any particular reason why we need to migrate to SVN/Git? Joegen On Monday, 31 May, 2010 03:20 AM, Meftah Tayeb wrote: > hi hfolk, > why we don't migrate opensipstack to svn or git? > > ------------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Meftah T. <tay...@gm...> - 2010-05-29 20:21:52
|
hi hfolk, why we don't migrate opensipstack to svn or git? |
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From: Joegen E. B. <jo...@op...> - 2010-05-21 01:10:13
|
The logs should be found in $(HOME)/OpenSBC_Data/logs. The wireshark capture seems to indicate an error in ICMP which could mean that the port has been closed by the remote end. Without actual logs to look at, the would be just a wild guess. Place a test call and attach your logs (zipped) together with your opensbc.ini (also in OpenSBC_Data) so I coulod give you a more concrete answer. On 05/21/2010 03:22 AM, Roberto Arias wrote: > Hi, I have a question.. > > Is there any error in OpenSipStack under Windows Server 2008? > > this works correctly? > > Thanks > > _________________________________________________________________ > Almacenamiento ilimitado, más cerca de lo que pensabas. Hotmail te 25 GB gratis. Registrate ahora > http://mail.live.com/ > ------------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Joegen B. <joe...@gm...> - 2010-05-21 00:35:38
|
For as long as you compiled it there, I don't see any potential issues. On 05/21/2010 03:22 AM, Roberto Arias wrote: > Hi, I have a question.. > > Is there any error in OpenSipStack under Windows Server 2008? > > this works correctly? > > Thanks > > _________________________________________________________________ > Almacenamiento ilimitado, más cerca de lo que pensabas. Hotmail te 25 GB gratis. Registrate ahora > http://mail.live.com/ > ------------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
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From: Roberto A. <rar...@ho...> - 2010-05-20 19:22:42
|
Hi, I have a question.. Is there any error in OpenSipStack under Windows Server 2008? this works correctly? Thanks _________________________________________________________________ Almacenamiento ilimitado, más cerca de lo que pensabas. Hotmail te 25 GB gratis. Registrate ahora http://mail.live.com/ |
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From: Dana J. <dj...@cs...> - 2010-05-20 18:35:46
|
Hi, I am new to openSBC and trying to familiarize myself with it. I am trying to get a simple call scenario to work using openSBC but for some reason it is not working. I am using two polycom (PVX) video clients and openSBC (installed on CentOS VM): The scenario is as follows : Polycom PVX1 (10.2.10.2) => openSBC (10.2.10.4) => Polycom PVX2 (10.2.10.13) I am using simple IP addressing for routing the call through openSBC, without (Registration). So for example Polycom PVX1 is +17034565000@10.2.10.2, Polycom PVX2 is john@10.2.10.13, and openSBC is 10.2.10.4 (port 5060) Both video clients are set to send calls via proxy ...10.2.10.4:5060 and this the openSBC.. PVX1's originating URI is entered in the openSBC's (Local-domain-accounts) tab (sip:+17034565000@10.2.10.2:5060), and PVX2's destination routing info is entered in the (B2BUA Routes) tab ( [sip:john] sip: 10.2.10.13:5060 ). I have openSBC on B2BUA mode (since I want to take advantage of B2BUA functionalities not proxy functionalities). I have also set the (SIP Transport) tab => main interface (blank), therefore it should default to using port 5060. (Trusted Domains) => Accept all calls (checked). When I make the calls I see on wireshark the following message: source destination Protocol Info. 10.2.10.2 (PVX1) 10.2.10.4 (openSBC) SIP/SDP INVITE sip:john@10.2.10.2 with session description 10.2.10.4 10.2.10.2 ICMP Destination Unreachable (Host Administravly Prohibited) Also, when trying to find the detailed logs (PTRACE) logs, which I have set to level 5 for detailed logging, I cannot seem to find them. I am not sure if they are being saved with a (ptrace) tag or something else, or even if they are being generated at all. I tried to follow the openSBC tab instructions by looking where PWLIB and OPAL libraries , but nothing found. Your feedback in these simple two questions is highly appreciated. Thanks Dana Jaff Systems Engineer |
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From: Joegen B. <joe...@gm...> - 2010-05-19 01:27:39
|
Hi Dana, I haven't tried inserting this header although if you are asking where is the best place in the code to insert headers for the outbound invite request, I would suggest doing it in SBCOutboundCall.cxx : SBCOutboundCall::OnSetupOutbound() assuming you are intending to do this using OpenjSBC code. If this is not what you are intending to do, feel free to clarify further. Joegen On 05/18/2010 11:04 PM, Dana Jaff wrote: > Hi, > > I was wondering if anyone has implemented insertion of Resource Priority > Header (RPH) for SIP emergency/priority calling. The RPH normally would > contain namespcaes such as (ets) and (wps) each with a relevant value that > corresponds to the users authorization level (Reference RFC 4421 sections > 4.5.2, 10.5, and 10.6). If so, how can this be achieved and implemented > using current software build. > > If not, we would like to work with the development group to achieve this. > Your feedback is greatly appreciated. > > Thanks > > Dana Jaff > CSC > > 15000 Conference Center Drive, Chantilly VA 20151 > Enforcement Security& Intelligence | O : 703-818-4135 | dj...@cs... | > www.csc.com > > This is a PRIVATE message. If you are not the intended recipient, please > delete without copying and kindly advise us by e-mail of the mistake in > delivery. > NOTE: Regardless of content, this e-mail shall not operate to bind CSC to > any order or other contract unless pursuant to explicit written agreement > or government initiative expressly permitting the use of e-mail for such > purpose. > ------------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
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From: Dana J. <dj...@cs...> - 2010-05-18 15:32:31
|
Hi, I was wondering if anyone has implemented insertion of Resource Priority Header (RPH) for SIP emergency/priority calling. The RPH normally would contain namespcaes such as (ets) and (wps) each with a relevant value that corresponds to the users authorization level (Reference RFC 4421 sections 4.5.2, 10.5, and 10.6). If so, how can this be achieved and implemented using current software build. If not, we would like to work with the development group to achieve this. Your feedback is greatly appreciated. Thanks Dana Jaff CSC 15000 Conference Center Drive, Chantilly VA 20151 Enforcement Security & Intelligence | O : 703-818-4135 | dj...@cs... | www.csc.com This is a PRIVATE message. If you are not the intended recipient, please delete without copying and kindly advise us by e-mail of the mistake in delivery. NOTE: Regardless of content, this e-mail shall not operate to bind CSC to any order or other contract unless pursuant to explicit written agreement or government initiative expressly permitting the use of e-mail for such purpose. |
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From: Thiago M. de S. <thi...@ho...> - 2010-02-25 17:00:51
|
Hi Joegen, I want to know how does a softphone would negotiate with the OpenSipStack what codecs to use and what codecs can be used. > Date: Thu, 25 Feb 2010 07:40:09 +0800 > From: joe...@gm... > To: ope...@li... > Subject: Re: [OpenSIPStack] How does the OpenSipStack handles with the codecs? > > Hi Thiago, > > Before I answer your question, I just want to know exactly in what > application context you are asking this question. Are you asking how > opensbc negotiates codecs using opensipstack or a how a softphone would > do negotiation using opensipstack? > > Joegen > > > Thiago Martins de Sousa wrote: > > Hi, > > > > I'm studying this library but I didn't understand how does it handles with the codecs and the audio streams. And where I can see what codecs this library supports? > > > > _________________________________________________________________ > > Quer compartilhar fotos com seus amigos? Conheça agora o Windows Live Fotos. > > http://www.eutenhomaisnowindowslive.com.br/?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=InfuseSocial > > ------------------------------------------------------------------------------ > > Download Intel® Parallel Studio Eval > > Try the new software tools for yourself. Speed compiling, find bugs > > proactively, and fine-tune applications for parallel performance. > > See why Intel Parallel Studio got high marks during beta. > > http://p.sf.net/sfu/intel-sw-dev > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > ------------------------------------------------------------------------------ > Download Intel® Parallel Studio Eval > Try the new software tools for yourself. Speed compiling, find bugs > proactively, and fine-tune applications for parallel performance. > See why Intel Parallel Studio got high marks during beta. > http://p.sf.net/sfu/intel-sw-dev > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel _________________________________________________________________ No Messenger você pode tranformar sua imagem de exibição num vídeo. Veja aqui! http://www.windowslive.com.br/public/tip.aspx/view/97?product=2&ocid=Windows Live:Dicas - Imagem Dinamica:Hotmail:Tagline:1x1:Mexa-se |
|
From: Joegen B. <joe...@gm...> - 2010-02-24 23:40:19
|
Hi Thiago, Before I answer your question, I just want to know exactly in what application context you are asking this question. Are you asking how opensbc negotiates codecs using opensipstack or a how a softphone would do negotiation using opensipstack? Joegen Thiago Martins de Sousa wrote: > Hi, > > I'm studying this library but I didn't understand how does it handles with the codecs and the audio streams. And where I can see what codecs this library supports? > > _________________________________________________________________ > Quer compartilhar fotos com seus amigos? Conheça agora o Windows Live Fotos. > http://www.eutenhomaisnowindowslive.com.br/?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=InfuseSocial > ------------------------------------------------------------------------------ > Download Intel® Parallel Studio Eval > Try the new software tools for yourself. Speed compiling, find bugs > proactively, and fine-tune applications for parallel performance. > See why Intel Parallel Studio got high marks during beta. > http://p.sf.net/sfu/intel-sw-dev > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
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From: Meftah T. <tay...@gm...> - 2010-02-24 21:33:05
|
hi, opensipstack have integrated media handling capability for codecs: G.711A/U, G.726, ILBC, speex and mayb G.729 with intel IPP thanks Le 24/02/2010 20:20, Thiago Martins de Sousa a écrit : > Hi, > > I'm studying this library but I didn't understand how does it handles with the codecs and the audio streams. And where I can see what codecs this library supports? > > _________________________________________________________________ > Quer compartilhar fotos com seus amigos? Conheça agora o Windows Live Fotos. > http://www.eutenhomaisnowindowslive.com.br/?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=InfuseSocial > ------------------------------------------------------------------------------ > Download Intel® Parallel Studio Eval > Try the new software tools for yourself. Speed compiling, find bugs > proactively, and fine-tune applications for parallel performance. > See why Intel Parallel Studio got high marks during beta. > http://p.sf.net/sfu/intel-sw-dev > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Thiago M. de S. <thi...@ho...> - 2010-02-24 19:20:48
|
Hi, I'm studying this library but I didn't understand how does it handles with the codecs and the audio streams. And where I can see what codecs this library supports? _________________________________________________________________ Quer compartilhar fotos com seus amigos? Conheça agora o Windows Live Fotos. http://www.eutenhomaisnowindowslive.com.br/?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=InfuseSocial |
|
From: Joegen B. <joe...@gm...> - 2009-12-20 22:42:42
|
Yes OpenSIPStack supports RFC 2833. If you have inherited from
SoftPhoneInterface, all you need to do is implement
virtual void Event_ReceivedDTMF(
const OString & eventInfo
) = 0;
to receive the DTMF. eventInfo should contain the DTMF digit sent by
the remote.
To send DTMF, simple call the function below
void SendRFC2833Tone(
const OString & tone,
int duration
);
where tone contains the digit(s). The duration should contain how long
the digits are played expressed in milliseconds.
~joegen
duong dao wrote:
> hi everybody,
>
> Does opensipstack support RFC2833? How do i send DTMF via RTP?
>
> Thanks you!
>
>
>
>
> ------------------------------------------------------------------------------
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
|
|
From: duong d. <duo...@ya...> - 2009-12-20 16:38:41
|
hi everybody,
Does opensipstack support RFC2833? How do i send DTMF via RTP?
Thanks you!
|
|
From: Meftah T. <tay...@gm...> - 2009-11-05 17:58:57
|
hello, please how do i compile atlsip in Visual Studio 2008? i don't see project for Visual Studio 2008 or 2005 i see only for VC7.10 but OpenSIPStack is for Visual Studio 2008 also i have a file included that don't existe, is not found: tpipv6.h thank you __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Meftah T. <tay...@gm...> - 2009-11-05 16:36:23
|
hello, thank you Joegen for your reply mayb i will implement G.722 for AtlSip and host it anyhere;) the G.722 implementation in freeswitch is very awesome thank you Joegen Baclor a écrit : > Hi Meftah, > > G.729 is already supported in ATLSIP using Voice Age. The configure > script should be able to detect it if it is installed in your system. > For G.722, you need to code it yourself. Simply base you > implementation on g729codec.*. You need to implement transcoder to and > fro g722 and PCM. and call OPAL_REGISTER_TRANSCODER when you have the > class ready. Example G729 register macro below. > > #define OPAL_REGISTER_G729() \ > OPAL_REGISTER_TRANSCODER(Opal_G729_PCM, OpalG729, OpalPCM16); \ > OPAL_REGISTER_TRANSCODER(Opal_PCM_G729, OpalPCM16, OpalG729); > > Once your codec is registered, it should already be part of the list of > codecs ATLSIP would support. > > Meftah Tayeb wrote: > >> hi all, >> how to add aditional Codecs to atlsip like G.722 and G.729? >> also what video codec is included with atlsip? >> thank you >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Meftah T. <tay...@gm...> - 2009-11-05 16:34:35
|
hi Joegen , Video is not fully needded;) dud;) Joegen E. Baclor a écrit : > I forgot to answer your question about video codec. No, there is no > video support for ATLSIP. This is purely a decision made due to the > fact that only H.261 codec can safely be used without any > licensing/patent issues. With its large bandwidth requirement, nobody > wants to use it nowadays. although if you really want delve into the > code, video support is right there lurking. Just dont expect much > support from this list when it comes to video. > > Meftah Tayeb wrote: > >> hi all, >> how to add aditional Codecs to atlsip like G.722 and G.729? >> also what video codec is included with atlsip? >> thank you >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Joegen E. B. <joe...@gm...> - 2009-11-04 23:52:15
|
I forgot to answer your question about video codec. No, there is no video support for ATLSIP. This is purely a decision made due to the fact that only H.261 codec can safely be used without any licensing/patent issues. With its large bandwidth requirement, nobody wants to use it nowadays. although if you really want delve into the code, video support is right there lurking. Just dont expect much support from this list when it comes to video. Meftah Tayeb wrote: > hi all, > how to add aditional Codecs to atlsip like G.722 and G.729? > also what video codec is included with atlsip? > thank you > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Joegen B. <joe...@gm...> - 2009-11-04 23:39:47
|
Hi Meftah,
G.729 is already supported in ATLSIP using Voice Age. The configure
script should be able to detect it if it is installed in your system.
For G.722, you need to code it yourself. Simply base you
implementation on g729codec.*. You need to implement transcoder to and
fro g722 and PCM. and call OPAL_REGISTER_TRANSCODER when you have the
class ready. Example G729 register macro below.
#define OPAL_REGISTER_G729() \
OPAL_REGISTER_TRANSCODER(Opal_G729_PCM, OpalG729, OpalPCM16); \
OPAL_REGISTER_TRANSCODER(Opal_PCM_G729, OpalPCM16, OpalG729);
Once your codec is registered, it should already be part of the list of
codecs ATLSIP would support.
Meftah Tayeb wrote:
> hi all,
> how to add aditional Codecs to atlsip like G.722 and G.729?
> also what video codec is included with atlsip?
> thank you
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> ------------------------------------------------------------------------------
> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day
> trial. Simplify your report design, integration and deployment - and focus on
> what you do best, core application coding. Discover what's new with
> Crystal Reports now. http://p.sf.net/sfu/bobj-july
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
|
|
From: Meftah T. <tay...@gm...> - 2009-11-04 23:11:41
|
hi all, how to add aditional Codecs to atlsip like G.722 and G.729? also what video codec is included with atlsip? thank you __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Joegen E. B. <joe...@gm...> - 2009-09-19 00:49:44
|
By default opensbc listens on port 9999 TCP for the admin. Make sure you punch a hole in your firewall to allow connections to this port. You may also want to allow 5060-5070 for signaling and 30000 - 35000 for RTP. Meenatchy Annamalai wrote: > Hi, > > I compiled and installed the openSBC in my linux macine. I am running > opensbc in daemon mode... openSBC is running.. but I am not able to > connect to the the admin page... pls help > > > Thanks & Regards, > *Meenatchy Annamalai > * > |
|
From: Meftah T. <tay...@gm...> - 2009-09-05 02:22:21
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hello Joegen , i tryed the property to true but same thing! is tryed with freeswitch 1.0.4 final, SipX and Asterisk! thanks Joegen Baclor wrote: > Hi, > > There is a sample application called OSSPhone in ATLSIP package. > DTMFAsRFC2833 should be set to true if you want to use RFC 2833 as > opposed to INFO. Make sure you call InitializeSIP() every time you > change a property. Have you tried sniffing the RTP packet using > ethereal? It should very much evident in ethreal if ATLSIP is or is not > sending the DTMF properly. > > Joegen > > Meftah Tayeb wrote: > >> hello all >> i'm trying DTMF using atlsip with asterisk >> i dialed *98 for voicemail but i can't dial my extension / password, >> why? is kype saying mailbox and password >> i see a property: DTMFAsRFC2833 of type byte >> how to use it? >> any sample? >> thanks! >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
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From: Joegen B. <joe...@gm...> - 2009-09-01 04:09:21
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Hi, There is a sample application called OSSPhone in ATLSIP package. DTMFAsRFC2833 should be set to true if you want to use RFC 2833 as opposed to INFO. Make sure you call InitializeSIP() every time you change a property. Have you tried sniffing the RTP packet using ethereal? It should very much evident in ethreal if ATLSIP is or is not sending the DTMF properly. Joegen Meftah Tayeb wrote: > hello all > i'm trying DTMF using atlsip with asterisk > i dialed *98 for voicemail but i can't dial my extension / password, > why? is kype saying mailbox and password > i see a property: DTMFAsRFC2833 of type byte > how to use it? > any sample? > thanks! > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
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From: Meftah T. <tay...@gm...> - 2009-08-30 03:10:18
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hello all i'm trying DTMF using atlsip with asterisk i dialed *98 for voicemail but i can't dial my extension / password, why? is kype saying mailbox and password i see a property: DTMFAsRFC2833 of type byte how to use it? any sample? thanks! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
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