I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close the connection but acts like there is still someone on the otherside. (Logging shows dat de Zap/channel has cleared, but not the SIP/channel) I use Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 Any help would be greatly appreciated... Ramon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050107/d9c6fcb5/attachment.htm
Ramon Peek wrote:> - When I pickup a call from another set, the *8 code keeps being > displayed in my screen (Snom 220). > I would like it to show the phonenumber of the person calling me.This is correct. You are placing a call to *8 which just happens to connect you to caller. As far as your phone is concerned it is talking to someone at *8.
Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira
Hi, With Gnugk, make sure the proxy mode is not enabled if you want voice to pass directly from endpoints. Regards Lamine ----- Original Message ----- From: "Joao Pereira" <joao.pereira@fccn.pt> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming> Hi > When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) > are just used for signaling, but the call streaming passes from theendpoint> directly to Asterisk, isnt it? Or does the streming passes from the > Endpoint to SER and then to the Asterisk? > > Thanks > Joao Pereira > > > > _______________________________________________ > 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 >
Ok, then I guess the way we use SER and GNUGK to redirect calls to Asterisk makes the diference. If we are using them as proxy, the stream will pass through them, if we dont use proxy, they will be used just for signaling. Joao ----- Original Message ----- From: "Mamadou Lamine KA" <lamineka@chaka.sn> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Friday, January 07, 2005 10:50 AM Subject: Re: [Asterisk-Users] Signaling / Streaming> Hi, > With Gnugk, make sure the proxy mode is not enabled if you want voice to > pass directly from endpoints. > Regards > Lamine > > ----- Original Message ----- > From: "Joao Pereira" <joao.pereira@fccn.pt> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Friday, January 07, 2005 10:21 AM > Subject: [Asterisk-Users] Signaling / Streaming > > > > Hi > > When I forward calls from SER (or GNUGK) to Asterisk, the SER ( orGNUGK)> > are just used for signaling, but the call streaming passes from the > endpoint > > directly to Asterisk, isnt it? Or does the streming passes from the > > Endpoint to SER and then to the Asterisk? > > > > Thanks > > Joao Pereira > > > > > > > > _______________________________________________ > > 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 >
Yes, This mode is generally used when some endpoints have private addresses behind a NAT while others have public addresses. In this case all the traffic passes through the GK. Take a look at paragraph related to Proxy at http://www.gnugk.org/gnugk-manual-4.html#ss4.2 Lamine ----- Original Message ----- From: "Joao Pereira" <joao.pereira@fccn.pt> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Friday, January 07, 2005 11:18 AM Subject: Re: [Asterisk-Users] Signaling / Streaming> Ok, > then I guess the way we use SER and GNUGK to redirect calls to Asterisk > makes the diference. > If we are using them as proxy, the stream will pass through them, if wedont> use proxy, they will be used just for signaling. > > Joao > > > > ----- Original Message ----- > From: "Mamadou Lamine KA" <lamineka@chaka.sn> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Friday, January 07, 2005 10:50 AM > Subject: Re: [Asterisk-Users] Signaling / Streaming > > > > Hi, > > With Gnugk, make sure the proxy mode is not enabled if you want voice to > > pass directly from endpoints. > > Regards > > Lamine > > > > ----- Original Message ----- > > From: "Joao Pereira" <joao.pereira@fccn.pt> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <asterisk-users@lists.digium.com> > > Sent: Friday, January 07, 2005 10:21 AM > > Subject: [Asterisk-Users] Signaling / Streaming > > > > > > > Hi > > > When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or > GNUGK) > > > are just used for signaling, but the call streaming passes from the > > endpoint > > > directly to Asterisk, isnt it? Or does the streming passes from the > > > Endpoint to SER and then to the Asterisk? > > > > > > Thanks > > > Joao Pereira > > > > > > > > > > > > _______________________________________________ > > > 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 > > > > _______________________________________________ > 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 >
> I know that my phone displays *8 because I dailed that. > But it's definitly not what I would want, or most other people. > Any other ordinary PBX would show the CID of the caller, but because thisis a SIP-based system we get this problem.> I was thinking more in line of an alternate call-pickup procedure torealize this option.> My idea would be: > > exten => *8,1,SetVar(PICKEXT=${CALLERIDNUM}) > exten => *8,2,HangUp > exten => *8,3, Here come the lines that will deflect the call to $PICKEXT > > Why deflect?, well when a call is deflected CID information is alsotransferred.> > The effect would be that when dialing *8 the connection would be closed,but immediatly after that your phone will ring showing you all the information you need.. even before pickup.> > You could call this function remote deflecting??? > > This function does not exist in * as far as I know, but perhaps there issome work-around to this???> > Anyone?!??!I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close the connection but acts like there is still someone on the otherside. (Logging shows dat de Zap/channel has cleared, but not the SIP/channel) I use Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 Any help would be greatly appreciated... Ramon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050107/796a2627/attachment.htm
Hi, can someone tell be about some good and free softphones? Are they easy to use by non-tecnical users? Can someone share their experience about the implementation of VoIP softphones in a company?.... because usualy people dont want to make changes in the way they work.... I would like to know a way to convince peaple in my company to use them. Thanks Joao Pereira