[asterisk-users] Multiple Meet me conferences
Hello, I was wondering if it is possible within Asterisk to be in many meetme conferences at the same time. This would be sort of broadcasting over all the conferences at once. Thanks, Denis ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Timeout issues
Hello, I have a softphone which I am using with Asterisk. Sometimes when I place a call it works fine and sometimes the SipListener comes back with a timeout. The timeout is a Retransmission timeout and it seems to be occurring when the INVITE is sent. The thing is about 70% of the time it works, but the other 30% or so it comes back with the timeout message. Is there any reason why it does this occasionally? I am not sure what to look for or how to get rid of these timeouts. Any help or advice would be appreciated. Thanks, Denis ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Softphone RTP Session Start-up Delay
Thanks for the reply. I actually found the problem by inserting print statements in the code and checking which operations took the most time. I found that one line was taking a long time to run. The line was the following RTPManager.addtarget(destAddress) After googling this for a while, I found a solution on a JMF site which recommended a minor change in the way that the destAddress is obtained. After making this change the delay was removed. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James FitzGibbon Sent: Wednesday, September 19, 2007 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone RTP Session Start-up Delay On 9/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: here because we are actually specifying the IP Address of the Asterisk server, but I am willing to try anything to fix this problem. The two user pc's are setup on workgroups, so I do not believe that there is a domain available that can be entered in the hosts file. Could the DNS still be the issue? If not, would anyone be able to suggest any other possible problems that may be causing this delay. It wasn't the same magnitude (more like 4 seconds for me), but I had an issue where the default eyeBeam (the commercial version of X-Lite) install was imposing a delay when a call first came in while it attempted to contact a non-existent STUN server. When I removed the STUN server setting, call setup was immediate. Might be worth looking at. Have you done a trace on the PC where the softphone is running (without a filter) to see what network packets are flying at the time the call setup happens? -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Softphone RTP Session Start-up Delay
Hi, Thanks very much for your reply. I would like to add some information which may provide a little more clarification on this matter. The LAN network that we presently have consists of the Asterisk PC and two User PC's (This network is not connected to the internet). To confirm that Asterisk/Trixbox operated correctly we installed an X-Lite phone on each user pc. We specified the IP Address of the Asterisk machine for the domain in the properties of the X-Lite. These X-Lites worked well, having no delay at any point in the process from when the call is made, up to the audio conversation. Unfortunately, the X-Lite phone is not open-source, so we do not have the code available to us. We then obtained the Jain-SIP phone, which is an open-source SIP softphone. As done in the X-Lite, the Asterisk IP Address is specified for the outbound proxy or the domain. We are now able to establish an audio conversation except for the fact that the RTP session takes about 20 seconds to setup, as mentioned before. I am not sure if the DNS issue comes into play here because we are actually specifying the IP Address of the Asterisk server, but I am willing to try anything to fix this problem. The two user pc's are setup on workgroups, so I do not believe that there is a domain available that can be entered in the hosts file. Could the DNS still be the issue? If not, would anyone be able to suggest any other possible problems that may be causing this delay. Thanks in advance for the help, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anselm Martin Hoffmeister Sent: Monday, September 17, 2007 4:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone RTP Session Start-up Delay Am Montag, den 17.09.2007, 15:50 -0400 schrieb [EMAIL PROTECTED]: Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would drastically reduce the delay but there was no change. I also tried a number of values for the minimum threshold and this did not change the amount of delay either. Would anyone have an idea of why this delay is occurring and possibly how to reduce it? Hello Denis, delays in that magnitude (20 seconds or about) may be related to DNS issues - like trying to resolve a hostname, or trying to find a hostname for an IP address. You could try to add all relevant IPs to the /etc/hosts file (or C:\windows\system32\drivers\etc\hosts), like 192.168.0.2 host2 and see wether that helps. Regards, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. *http://*www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Softphone RTP Session Start-up Delay
Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would drastically reduce the delay but there was no change. I also tried a number of values for the minimum threshold and this did not change the amount of delay either. Would anyone have an idea of why this delay is occurring and possibly how to reduce it? Any advice would be greatly appreciated, Thanks, Denis ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Chan_sip Entry
Hello, Yes, I also believe that this is some sort of codec issue. Here is my sip.conf file: [201]?xml:namespace prefix = o ns = urn:schemas-microsoft-com:office:office / type=friend ;secret=201 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend ;secret=202 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Note: The secret is commented out so that there is no authentication when registering with the Jain-Sip phones. Thanks, -Original Message- From: Gerald A [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 11, 2007 5:12 PM To: Kutman [EMAIL PROTECTED](Mat) DAEPM(RCS)@Ottawa-Hull Subject: Re: [asterisk-users] Chan_sip Entry Hi again, On 9/11/07, [EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: Oooh, format changed to 2. Usually this is a codec selection problem. Are both Jain's the same version? Maybe posting your sip.conf for the phones might help. Thanks, Gerald. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] Chan_sip Entry
Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: Oooh, format changed to 2. Would anyone know why this is occuring one way and not the other, and more importantly, how would I fix this. After some examination I see that when I send the OK to the INVITE, this SDP body should have a 0 for the codec which is ulaw. When this Ok message gets to the other pc after going through asterisk it seems like asterisk adds a codec because the SDP body now contains the codecs 0 and 3. I believe the problem has something to do with this but I am not sure why it would work one way but not the other. Any help would be greatly appreciated. Thanks very much, Denis Kutman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 audioconversationbetweensoftphonesthrough Asterisk
Hello, Looks like I have been able to get the jain-sip-phone to work. The problem seemed to have been an sdpFactory.createconnection call. It was passing one parameter, which was the IP Address. I had to change this to the call with three parameters (ie: sdpFactory.createconnection(IN, IP4, etc). This is because by default eclipse was setting the address type to IPV4, which didn't seem to work. Such a minor issue, but it wasn't easy to find. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 28, 2007 9:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Can't create audioconversationbetweensoftphonesthrough Asterisk Hello, I do not think that the presence bit will be crucial to our application. Thanks for your help. I will keep you posted if I get any progress. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A Sent: Monday, August 27, 2007 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't create audio conversationbetweensoftphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Glad to hear you were able to get some traction with the voice calling. Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. Keep me updated on your progress, and if you need any assistance, give me a shout. 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 conversationbetweensoftphonesthrough Asterisk
Hello, I do not think that the presence bit will be crucial to our application. Thanks for your help. I will keep you posted if I get any progress. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A Sent: Monday, August 27, 2007 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't create audio conversationbetweensoftphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Glad to hear you were able to get some traction with the voice calling. Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. Keep me updated on your progress, and if you need any assistance, give me a shout. 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, 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 betweensoftphonesthrough Asterisk
Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A Sent: Monday, August 27, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't create audio conversation betweensoftphonesthrough 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
[asterisk-users] Can't create audio conversation between softphones through 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=z9hg4bk583c7ea7c1a8da8576583356f821c9c2 firstLine=SIP/2.0 488 Not acceptable here callId=[EMAIL PROTECTED] debugLine=0 ![CDATA[SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2;received=192.168.1.251 From: sip:[EMAIL PROTECTED];tag=2085 To: sip:[EMAIL PROTECTED];tag=as2f851644 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 Has anyone ever tried using these Jain-sip-applet-phones and got them to work? I have read up on these errors, and it looks like the 489 error doesn't like the SUBSCRIBE request, while the 488 error doesn't seem to accept the INVITE request made. I am not sure if this is a problem with Asterisk, incompatibility between Asterisk and the phones, or just the phones. Any thoughts that may help me resolve these issues would be greatly appreciated. Thanks very much, Denis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
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
[asterisk-users] Asterisk Message Logs
Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through an Asterisk server. I can't seem to get communication between the two phones. Does anyone have any experience using these open-source Jain-sip-applet-phones? Thanks, Denis ___ --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] Asterisk Message Logs
Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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 ___ --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] Asterisk Message Logs
Yes, any output from the console logs. I tried viewing the full file and it looks like it's what I was looking for. Thanks for the help. Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Jones Sent: Thursday, August 23, 2007 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs [EMAIL PROTECTED] wrote: Thanks for your reply. I have previously looked at the logger.conf file. I see that the various types of information can be logged in different ways. After setting the various information types with whatever I want logged, is it possible to save the actual logs to a file (ie: As the messages are bring printed, save them all to a file to be viewed later). What do you mean by actual logs? Console (CLI) output? Brian. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jared Smith Sent: Thursday, August 23, 2007 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Message Logs On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote: Is it possible to print the Asterisk message logs to a file, or is this already done? You want to look at the logger.conf configuration file, and see how your Asterisk system is set to log the various types of information (such as debug messages, verbose messages, DTMF messages, etc.) are logged. After changing logger.conf, you can type logger reload at the Asterisk CLI to make the changes take effect. ___ --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 ___ --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
[asterisk-users] Disabling Asterisk Authentication
Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a 401 Unauthorized error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Thanks very much, Denis Kutman ___ --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
[asterisk-users] Jain-Sip-Applet-Phone with Asterisk
Hello, I have the Jain-Sip-Applet-Phone installed on two machines in a small LAN network. These machines are connected through an Asterisk Server (Using Trixbox). I run the phone as an application on both machines through Eclipse and I am able to log on as a user with one of the extensions that I use within Asterisk on each machine (extensions 201 and 202 in this case). When I try to add a contact to my list (the other user machine) from one of the machines, I get the following message, which looks like I am having a problem with authentication: ECLIPSE CONSOLE WINDOW ![CDATA[SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.252:7933;branch=z9hG4bKf87a334e3afb041a4b7783d409b6c95d;received=192.168.1.252 From: sip:[EMAIL PROTECTED];tag=973 To: sip:[EMAIL PROTECTED];tag=as72bce758 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY WWW-Authenticate: Digest algorithm=MD5,realm=asterisk,nonce=62a12192 Content-Length: 0 I have defined two users in my sip.conf file as shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device 201 [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device 202 Would anyone know where I am going wrong? Is it possible to remove this authentication requirement? Thanks in advance, Denis Kutman ___ --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
[asterisk-users] Asterisk RTP bridging
Hello, I have a small LAN network connected through an Asterisk Server (Trixbox). I was looking to create my own custom made softphones, and I have been looking into how to transmit and receive via RTP. Would anyone know how the Asterisk RTP bridging works, and if there is any documentation on it? How is the RTP stream routed through the Asterisk server? Do I just give it the endpoints and then the audio call is transmitted directly between the two machines? Any help would be appreciated. Thanks very much, Denis ___ --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
[asterisk-users] Sending live audio in Asterisk
Hello, I am trying to create a Java GUI that will interact with an Asterisk Server. This Java GUI will essentially be a custom made SIP softphone. I will most likely use the Asterisk-Java Live API to create the connection to the Asterisk server and to open a new call. Then, I plan to use the JAIN SIP API to initialize the session and the JMF to send the audio streams via RTP when the two users are connected in a call. I had two questions about this type of system: 1. I believe I have a good idea of the overall process of opening a SIP session and streaming live audio(phone conversations) via RTP, but is there any Asterisk- specific sources or examples that start a session via SIP and then transmit the audio via RTP all done through the Asterisk server? 2. I know there is an rtp.conf file which outlines the ports available for rtp transfer. How is the actual RTP transfer between users completed through the Asterisk server? I am looking to transmit live audio between the users through the Asterisk server once the call is connected. Thanks in advance, Denis Kutman ___ --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
[asterisk-users] Creating an SIP softphone
Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users' PC's once the calls are routed. I can see that the Asterisk-Java library can't be used for transmitting real-time audio(phone conversations). Would anyone have an idea about how to complete the application I am trying to make. To be clear, I am creating a custom softphone, but can't find much information on how to create the audio transmission. Could anyone provide me with some advice on how to complete this type of softphone. I noticed that there is a JAIN(SIP) API that can be used with java, but I would need more information or examples on how this can be used for my application to use this API. I would prefer to use the SIP protocol, since it seems like its the most common. Thanks in advance, Denis ___ --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
[asterisk-users] RTP Session Streaming
Hello, I am trying to transmit and receive sound over IP using the Java Media Framework(JMF) RTP. I was wondering if its possible to create an RTP Stream from my own computer and assign it to a URL. If anyone knows how I would do this, could they point me to some instructions or an example. So far, I have some sample code which compiles and creates the player, but it can't seem to realize the player, due to the URL that is given in the example. I'm guessing I need to create my own RTP stream to test this voice receiver. It would be nice if I could write the code for a voice transmitter and receive the RTP Stream from the transmitter using my voice receiver but I am not sure if this is possible on one PC. Would anyone be able to give me some advice on this matter. Thank you, Denis ___ --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
