Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 "SDP body missing" message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints?
Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 "SDP body missing" message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints?
What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired? Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose "use authentication", fill in the blanks, I found also putting the "realm" in made it work correctly. Hope that helps in some way. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 15:17 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ahead SIPPS and Asterisk Has anyone gotten Ahead's SIPPS softphone to work with Asterisk? I get it to register, but when I dial into a number as soon as voice is to be connected I get a Warning 399 "SDP body missing" message and then a BYE disconnecting the call. The setup I have works great with Xten's x-pro, but can't get it to work with SIPPS. Any hints? _______________________________________________ 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'll try to look over my config again. Not sure I put a "realm" in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -----Original Message----- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 -0000 From: "Craig Waddington" <craig@xmbsystems.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose "use authentication", fill in the blanks, I found also putting the "realm" in made it work correctly. Hope that helps in some way.
I'll try to look over my config again. Not sure I put a "realm" in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -----Original Message----- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 -0000 From: "Craig Waddington" <craig@xmbsystems.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose "use authentication", fill in the blanks, I found also putting the "realm" in made it work correctly. Hope that helps in some way.
Your problem is what I experience with Messenger, when I call it. Unfortunately I never bothered trying to work out the problem. I like the SIPPS phone features, but it is ugly. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Carlton J. O'Riley Sent: 05 March 2004 18:48 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk I'll try to look over my config again. Not sure I put a "realm" in, but everything else seemed fine. I get the acquired message and I see the SIP messages flowing on the Asterisk server, but once any sound needs to be sent, it dies. I'm using g711ulaw and I was calling into an announcement menu to test that I have setup on the server. The entry in my sip.conf is exactly what I use for X-Pro, so that is why I am confused by the missing SDP message. -----Original Message----- Message: 8 Subject: RE: [Asterisk-Users] Ahead SIPPS and Asterisk Date: Fri, 5 Mar 2004 18:25:05 -0000 From: "Craig Waddington" <craig@xmbsystems.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com What sort of phone are you trying to call? I use SIPPS and * and it works fine, it just wont work when you call Windows Messenger for some reason. I can call X-lite, POTS, GS phones no problems. I use the same config as X-lite, in SIPPS if you click on the spanner or press F9 to go into configuration and click on network. You should see aquired?=20 Click modify and ensure Gateway is ticked and you have the IP of your server in there, for some reason it defaults as redirect. On mine I leave Dial prefix blank. Click modify again then Ok, which will take you to the config section for the user, choose "use authentication", fill in the blanks, I found also putting the "realm" in made it work correctly. Hope that helps in some way. _______________________________________________ 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