Klaus Hueske
2003-Apr-24 04:29 UTC
[Asterisk-Users] Outgoing SIP Call to unregistered Users
Hi! I'm using asterisk with a few kphone SIP-Clients. The registration process seems quite OK. But there are some problems: Calling other registered users is possible, but the rtp-stream is not reaching the right port, so you can hear nothing. In ethereal you can see, that the SIP/SDP fields addresses different ports at each client, so client A sends to port 32000 but client B listens on port 32002. One solution for this problem ist to use the canreinvite=no statement in sip.conf, but in this case every rtp-packet is going through asterisk. I think, only the SIP/SDP packets should go through asterisk and the voicetraffic direct from client A to client B. May be, I'm wrong about that, please correct me in that case. Another problem is calling SIP users that are not registered to asterisk. Giving kphone the address sip:name@anyhost causes asterisk to search for the extension name, but there is no such extension. Are there ways to say asterisk, that these calls should only be forwarded to the given host? I hope, somebody could write something about that. Thanks Klaus
Jamin W. Collins
2003-Apr-24 07:40 UTC
[Asterisk-Users] Outgoing SIP Call to unregistered Users
On Thu, Apr 24, 2003 at 01:29:27PM +0200, Klaus Hueske wrote:> Another problem is calling SIP users that are not registered to > asterisk. Giving kphone the address sip:name@anyhost causes asterisk > to search for the extension name, but there is no such extension. Are > there ways to say asterisk, that these calls should only be forwarded > to the given host?Last I checked, no. To do this, I put a SIP proxy in front of the * application on the same box. All inbound SIP traffic is handled by the SIP proxy and passed on to * as needed. -- Jamin W. Collins This is the typical unix way of doing things: you string together lots of very specific tools to accomplish larger tasks. -- Vineet Kumar
> Last I checked, no. To do this, I put a SIP proxy in front of the * > application on the same box. All inbound SIP traffic is handled by the > SIP proxy and passed on to * as needed.I thought * was a SIP proxy?? As well as everything else.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
Which SIP proxy do you use?? I would like to try it out..> On Thu, Apr 24, 2003 at 03:33:26PM +0000, WipeOut . wrote: > > > Last I checked, no. To do this, I put a SIP proxy in front of the * > > > application on the same box. All inbound SIP traffic is handled by > > > the SIP proxy and passed on to * as needed. > > > > I thought * was a SIP proxy?? As well as everything else.. > > Not by any definition that I'm aware of. I could be wrong, but last > time I tried to use * as a SIP proxy I ran into a problem with any > non-registered destination. Since then I've used an actual dedicated > Linux SIP proxy application on the same box. Works like a charm. > > -- > Jamin W. Collins > > This is the typical unix way of doing things: you string > together lots of very specific tools to accomplish larger tasks. > -- Vineet Kumar > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
Jamin W. Collins
2003-Apr-24 11:54 UTC
[Asterisk-Users] Outgoing SIP Call to unregistered Users
On Thu, Apr 24, 2003 at 04:50:57PM +0000, WipeOut . wrote:> Which SIP proxy do you use?? I would like to try it out..SER: http://iptel.org/ser/ -- Jamin W. Collins This is the typical unix way of doing things: you string together lots of very specific tools to accomplish larger tasks. -- Vineet Kumar
Ask Bjørn Hansen
2003-Apr-24 13:40 UTC
[Asterisk-Users] Outgoing SIP Call to unregistered Users
On Thursday, Apr 24, 2003, at 07:40 US/Pacific, Jamin W. Collins wrote:> Last I checked, no. To do this, I put a SIP proxy in front of the * > application on the same box. All inbound SIP traffic is handled by the > SIP proxy and passed on to * as needed.Which SIP proxy do you use? - ask -- http://www.askbjoernhansen.com/
On Thu, 24 Apr 2003, William Zhang wrote:> Anyone knows if there is any Softphone that use IAX?Gnophone http://www.gnophone.com/ -- Jon Stockill jon@stockill.org.uk
Do you know of a Windows version that works with Asterisk? Michael Rose,>On Thu, 24 Apr 2003, William Zhang wrote: > >> Anyone knows if there is any Softphone that use IAX? > >Gnophone > >http://www.gnophone.com/ > >-- >Jon Stockill >jon@stockill.org.uk