Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem
I can tell you that you are not alone. It's an issue I believe with Firefly, and not in your configurations.> > Message: 8 > Date: Fri, 15 Oct 2004 13:06:17 +0200 > From: Willem de Groot <willem@byte.nl> > Subject: [Asterisk-Users] FireFly w/ SIP > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <416FAF29.7070803@byte.nl> > Content-Type: text/plain; charset=us-ascii; format=flowed > > Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? > > It works in IAX mode, but in SIP mode I am unable to hear anything (no > dialtone, no voice). I am able to setup a conversation with another SIP > phone though (Xlite, Grandstream) and the other side can hear the > FireFly user just fine (both sides using g711u). > > I tried different PC's with different audio hardware. They all work fine > using FireFly in IAX mode and using other softphones, so I guess it must > be related so FireFly in SIP mode. > > This is my SIP config: > > [201] > type=friend > host=dynamic > dtmfmode=rfc2833 > context=sip > canreinvite=yes > > FireFly is also configured for rfc2833 dtmf. > > Thanks for any suggestions! > Willem > > >
I use FireFly w/ SIP all day long and it works great, except for the SIP registration interval which I was just told will be fixed in next weeks version. Are you using GSM or g711u? [remote-laptop] context=remoteusers type=friend username=remote-laptop secret=hiddenfromlist qualify=yes host=dynamic canreinvite=no dtmfmode=inband nat=yes callerid="John Doe" <1235551212> accountcode=7499 amaflags=billing That's what I have in my sip.conf Then tell firefly to use the ip of your asterisk server as the Server. Give it the user id and password. Uncheck "disable registration" and check "Active" Always worked for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Willem de Groot Sent: Friday, October 15, 2004 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] FireFly w/ SIP Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? It works in IAX mode, but in SIP mode I am unable to hear anything (no dialtone, no voice). I am able to setup a conversation with another SIP phone though (Xlite, Grandstream) and the other side can hear the FireFly user just fine (both sides using g711u). I tried different PC's with different audio hardware. They all work fine using FireFly in IAX mode and using other softphones, so I guess it must be related so FireFly in SIP mode. This is my SIP config: [201] type=friend host=dynamic dtmfmode=rfc2833 context=sip canreinvite=yes FireFly is also configured for rfc2833 dtmf. Thanks for any suggestions! Willem _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
The best way for me or yourself to debug it is using ethereal (google for it) and debugview from www.sysinternals.com. I'm happy to help, so send the logs, the native transfer might be the issue. -Adam Willem de Groot wrote:> Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? > > It works in IAX mode, but in SIP mode I am unable to hear anything (no > dialtone, no voice). I am able to setup a conversation with another SIP > phone though (Xlite, Grandstream) and the other side can hear the > FireFly user just fine (both sides using g711u). > > I tried different PC's with different audio hardware. They all work fine > using FireFly in IAX mode and using other softphones, so I guess it must > be related so FireFly in SIP mode. > > This is my SIP config: > > [201] > type=friend > host=dynamic > dtmfmode=rfc2833 > context=sip > canreinvite=yes > > FireFly is also configured for rfc2833 dtmf. > > Thanks for any suggestions! > Willem > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Adam On UK keyboards ,I have to type a "?" to get a "#" into Firefly. The proper "#" key does nothing. If you are updating the code, perhaps you might look at this? Many thanks Peter -----Original Message----- From: Adam Hart [mailto:adam@teragen.com.au] Sent: 16 October 2004 07:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FireFly w/ SIP The best way for me or yourself to debug it is using ethereal (google for it) and debugview from www.sysinternals.com. I'm happy to help, so send the logs, the native transfer might be the issue. -Adam Willem de Groot wrote:> Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? > > It works in IAX mode, but in SIP mode I am unable to hear anything (no > dialtone, no voice). I am able to setup a conversation with another SIP > phone though (Xlite, Grandstream) and the other side can hear the > FireFly user just fine (both sides using g711u). > > I tried different PC's with different audio hardware. They all work fine > using FireFly in IAX mode and using other softphones, so I guess it must > be related so FireFly in SIP mode. > > This is my SIP config: > > [201] > type=friend > host=dynamic > dtmfmode=rfc2833 > context=sip > canreinvite=yes > > FireFly is also configured for rfc2833 dtmf. > > Thanks for any suggestions! > Willem > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you.