[asterisk-users] Critical - No audio issue with re-invite (wrong media address)
Hi all I faced the issue with canreinvite=yes in Asterisk (both 1.2.9.1 and 1.2.1.12). My diagram UA1(RFC1918) - ADSL1 (NAT) -- Asterisk (Public IP) (NAT)ADSL2 --- UA2 (RFC1918) When canreinvite=no in sip.conf for both UAs, everything is fine, RTP from both UAs are sent to Asterisk (can see using rtp debug) and both UAs can talk. When canreinvite=yes and nat=yes, I faced no audio issue. After debug, I found that, when Asterisk send re-invite to both UAs, both UAs reply their OK with c=their RFC1918 address, and Asterisk send that OK parameters to other UAs Asterisk send re-invite to UA1 with c=Public NATed IP of ADSL2 Asterisk send re-invite to UA2 with c=Public NATed IP of ADSL1 UA1 reply OK to Asterisk with c=RFC1918 of UA1 UA2 reply OK to Asterisk with c=RFC1918 of UA2 Asterisk send OK to UA2 with c= RFC1918 of UA1 (because it received the newer c= RFC1918 of UA1) Asterisk send OK to UA1 with c= RFC1918 of UA2 (because it received the newer c= RFC1918 of UA2) UA1 send RTP directly to c= RFC1918 of UA2 (it must be c=Public NATed IP of ADSL2 for correct audio flow) UA2 send RTP directly to c= RFC1918 of UA1 (it must be c=Public NATed IP of ADSL1 for correct audio flow) no audio at all Pls. help Brgds Hoa Sip.conf [general] context=default port=5060 bindaddr=0.0.0.0 realm= srvlookup=no disallow=all allow=g729 canreinvite=yes nat=yes [] type=friend host=dynamic username= secret= canreinvite=yes nat=yes callerid= allow=g729 [] type=friend host=dynamic username= secret= canreinvite=yes nat=yes callerid= allow=g729 extensions.conf [general] static=yes writeprotect=no exten = ,1,Answer exten = ,2,Dial(SIP/) exten = ,2,Hangup exten = ,1,Answer exten = ,2,Dial(SIP/) exten = ,2,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP reinvite still does not occour
Roger If transcoding happened (from G.729 to GSM, any to any) then Asterisk will issue no re-INVITE, for sure. Pls. change Disallow=all Allow=gsm (only one codec) Then test, you'll see it happen. Cheers Hoa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Schreiter Sent: Friday, June 30, 2006 8:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP reinvite still does not occour Hi, I have in my sip.conf disallow=all allow=alaw in order to avoid any codec problems disturbing reinvite. And of course I have: canreinvite=yes In extensions.conf there is only one Dial command. It has no qualifiers like t or T. Just Dial(SIP/[EMAIL PROTECTED]) Anyway, asterisk does not try to reinvite. asterisk tells -- Attempting native bridge of SIP/01234567 ... but in the debug output there no reinvite. Using tcpdump I can see, that the audio data are going via the asterisk box in the middle, not direct between the endpoints. Is there anything else, which can prevent a reinvite? dtmp-settings? nat-settings? Thanks for any hints! Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Background + Dial
Hi GL Pls. config MOH and use Dial command with m option. This will allow you execute Dial command while providing Music in the background. Hope it help Hoa Thai Duy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 27, 2006 5:01 PM To: asterisk-users Subject: [Asterisk-Users] Background + Dial Hi everybody, I try this : [incoming_from_fxo_card] exten = s,1,Answer() exten = s,2,Background(filename) exten = s,3,Dial($(INTERNAL_SIP_TEL)) But * wait the file is finish before make Dial to SIP channel. Background(filename) (from voip-info.org) = Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next commands (if any) execute. I want to Dial a SIP channel while playing sound and waiting for a digit from a ZAP channel. In other words, i want to make a interactive MoH while waiting for the SIP channel answer. Is it possible? Thanks a lot and excuse me for my poor english (I'll fix this in few months). GL - ALICE SECURITE ENFANTS - Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants, le contrôle parental d'Alice. http://www.aliceadsl.fr/securitepc/default_copa.asp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled- ugrent
Title: SIP Channel hangup problem with re-INVITE enabled - ugrent Does anyone on this list has idea? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hoa Thai DuySent: Thursday, June 22, 2006 2:50 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Cc: asterisk-dev@lists.digium.comSubject: [Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled- ugrent Hi List I have UAs registered with Asterisk and make outbound calls via ITSP1, everything is fine without re-INVITE. When people call 178, the actual number 112233445566 at ITSP1 network will be called. When UA or called telephone (112233445566) hang up, the call and associated channels are cleared. Sip.conf [general] canreinvite=no nat=no [ITSP1] type=peer host=A.B.C.D Extensions.conf exten = 178,1,Answer() exten = 178,n,Dial(SIP/[EMAIL PROTECTED],60) exten = 178,n,Hangup() However, when I enabled re-INVITE like below, the call still happen, people can talk with each other. If remote called telephone (112233445566) hang up, then the call is cleared. But if the Asterisk user (US) Softphone hang up first, the remote telephone still in talking mode (with no sound, of course). Sip.conf [ITSP1] type=peer host=A.B.C.D Canreinvite=yes Nat=yes In this case, when Asterisk user hang up and remote phone still not hang up, I do show like this Show channel verbose 0 active channels 0 active calls Sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message A.B.C.D 112233445566 14448d41170 00103/00104 unkn No (d) Rx: BYE CLI sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format unknown Theoretical Address: A.B.C.D:5060 Received Address: A.B.C.D:5060 NAT Support: Always Audio IP: W.X.Y.Z(local) Our Tag: as5436f254 Their Tag: caba969d04802f1091a1--558 SIP User agent: Asterisk Username: 112233445566 Peername: 112233445566 Original uri: sip:[EMAIL PROTECTED]:5060 Need Destroy: 2 Last Message: Rx: BYE Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060;transport=UDP DTMF Mode: rfc2833 SIP Options: (none) In this case, when Asterisk user hang up and remote phone still not hang up, there's still active SIP channel, which should be cleared when BYE received from any of peers. In Asterisk Console, I can see BYE from Asterisk user (UA Softphone) to Asterisk and OK from Asterisk to UA. But Asterisk DO NOT send BYE to ITSP1, which is wrong? Pls. advice Brgds Hoa ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Channel hangup problem with re-INVITE enabled - ugrent
Title: SIP Channel hangup problem with re-INVITE enabled - ugrent Hi List I have UAs registered with Asterisk and make outbound calls via ITSP1, everything is fine without re-INVITE. When people call 178, the actual number 112233445566 at ITSP1 network will be called. When UA or called telephone (112233445566) hang up, the call and associated channels are cleared. Sip.conf [general] canreinvite=no nat=no [ITSP1] type=peer host=A.B.C.D Extensions.conf exten = 178,1,Answer() exten = 178,n,Dial(SIP/[EMAIL PROTECTED],60) exten = 178,n,Hangup() However, when I enabled re-INVITE like below, the call still happen, people can talk with each other. If remote called telephone (112233445566) hang up, then the call is cleared. But if the Asterisk user (US) Softphone hang up first, the remote telephone still in talking mode (with no sound, of course). Sip.conf [ITSP1] type=peer host=A.B.C.D Canreinvite=yes Nat=yes In this case, when Asterisk user hang up and remote phone still not hang up, I do show like this Show channel verbose 0 active channels 0 active calls Sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message A.B.C.D 112233445566 14448d41170 00103/00104 unkn No (d) Rx: BYE CLI sip show channel [EMAIL PROTECTED] * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format unknown Theoretical Address: A.B.C.D:5060 Received Address: A.B.C.D:5060 NAT Support: Always Audio IP: W.X.Y.Z(local) Our Tag: as5436f254 Their Tag: caba969d04802f1091a1--558 SIP User agent: Asterisk Username: 112233445566 Peername: 112233445566 Original uri: sip:[EMAIL PROTECTED]:5060 Need Destroy: 2 Last Message: Rx: BYE Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060;transport=UDP DTMF Mode: rfc2833 SIP Options: (none) In this case, when Asterisk user hang up and remote phone still not hang up, there's still active SIP channel, which should be cleared when BYE received from any of peers. In Asterisk Console, I can see BYE from Asterisk user (UA Softphone) to Asterisk and OK from Asterisk to UA. But Asterisk DO NOT send BYE to ITSP1, which is wrong? Pls. advice Brgds Hoa ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Multi Call Generation
Very famous in SIP world is SIPP - http://sipp.sourceforge.net/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Lateef Sent: Thursday, June 22, 2006 6:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Multi Call Generation Hi all, Is there any such as tools for multi call generation to test, how much call can be done via Asterisk? _ Best Regards, --- Abdul Lateef Nepal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
