Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk
Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Thanks, Denis -Original Message- From: Gerald A [mailto:[EMAIL PROTECTED] Sent: Monday, August 27, 2007 9:30 AM To: Kutman [EMAIL PROTECTED](Mat) DAEPM(RCS)@Ottawa-Hull Subject: Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks for the reply. I have a small LAN network which I have connected with an Asterisk server. My Asterisk box and the user pc's are connected through a LAN switch. This network is not connected to the internet. The UNREACHABLE message does seem to point to what you mentioned below (Asterisk not being able to ping the phones), which seems weird to me. When I use X-Lite softphones on those user pc's, I can connect them to the Asterisk server fine and make calls. The subscription occurs when I try to add another contact(In the same LAN network) from one of the user pc's. I am attaching the console results that I get within Eclipse when I run this softphone. Ok, one more silly question -- might it be possible to do this with IAX? (I tend to lean on IAX for things, as it's more versitile and robust, if not so widely deployed). I'm not sure exactly what you are trying to accomplish, so I'm focusing on the questions you are having issues with. A bit of context might show up as another solution, though -- if you are able to provide it. I don't have time right now to dig through the traces, but I have a related question. Have you ever got a call to go through dialling from one Jain client to the other, without the subscription? My gut feeling is that there might be a basic config issue with the Jain client that is causing an issue, as what you want to do doesn't sound too difficult. Thanks, Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk
Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important classes are more or less written already. In short, my goal is to create a custom softphone GUI interface. I am using this Jain-sip-phone as an example, so that I could learn the SIP protocol/RTP transmission better. The reason I asked is because IAX works better through firewalls and is easier to troubleshoot. It's not as widely deployed as SIP, but it does work around some major things that SIP makes harder. I'm not sure of the quality or lineage of the JAIN application code, so can't comment if it's a good jumping off point. I have not really started altering much of the code yet because I was trying to see if it would run as is, so I have not tried dialing the Jain clients without a subscription. I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Subscription is used for presence. It can be used in an IM type app, or to light up a button on a phone when someone is busy. It shouldn't be needed to exchange a call though, and if you can do it without the subscription piece then it could help to pin down the issue you are having. (It might be _just_ the subscribe that is having an issue). I should have time later this afternoon to check your traces, and I'll try and give Jain a kick. Thanks, Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk
This is the full log that I get after my trial run: Aug 24 14:15:51 VERBOSE[3710] logger.c: -- Registered SIP '202' at 192.168.1.250 port 9810 expires 120 Aug 24 14:15:52 VERBOSE[3710] logger.c: -- Registered SIP '201' at 192.168.1.251 port 8529 expires 120 Aug 24 14:15:55 NOTICE[3710] chan_sip.c: Peer '202' is now UNREACHABLE! Last qualify: 0 Aug 24 14:15:56 NOTICE[3710] chan_sip.c: Peer '201' is now UNREACHABLE! Last qualify: 0 Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: Setting NAT on RTP to 0 Aug 24 14:16:10 WARNING[3710] chan_sip.c: Invalid host in c= line, 'IN IPV4 192.168.1.251' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: SIP message could not be handled, bad request: b475318241b3dca93128681e6f079093 192.168.1.251 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, August 24, 2007 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a 489 Bad Event SIP error shown below in red) [EMAIL PROTECTED] has been added to your contacts. null send request: SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721756281 isSender=true transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 callId=[EMAIL PROTECTED] firstLine=SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Length: 0 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721756281 isSender=false statusMessage=normal processing transactionId=z9hg4bk361290cad5885dbc4a03b5951cc85585 firstLine=SIP/2.0 489 Bad Event callId=[EMAIL PROTECTED] debugLine=0 ![CDATA[SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251 From: sip:[EMAIL PROTECTED];tag=8505 To: sip:[EMAIL PROTECTED];tag=as2cf724e9 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 3. Try to call that contact to create an audio conversation.(Get a 488 Not Acceptable Here SIP error shown below in blue) Get chat session: [EMAIL PROTECTED] Chat Session added: [EMAIL PROTECTED]:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In conversation with [EMAIL PROTECTED],resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true] 5 4 3 0 send request: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 v=0 o=201 908031 909400 IN IP4 192.168.1.251 s=- c=IN IPV4 192.168.1.251 t=0 0 m=audio 2448 RTP/AVP 5 4 3 0 message from=192.168.1.251:8386 to=192.168.1.10:5060 time=1187721758593 isSender=true transactionId=z9hg4bk583c7ea7c1a8da8576583356f821c9c2 callId=[EMAIL PROTECTED] firstLine=INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 debugLine=0 ![CDATA[INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: sip:[EMAIL PROTECTED]:8386;transport=udp Content-Type: application/sdp Content-Length: 114 ]] /message message from=192.168.1.10:5060 to=192.168.1.251:8386 time=1187721758609 isSender=false statusMessage=normal processing transactionId
