[asterisk-users] Billing
Hi I have the following topology: Sever A IAX2 - SERVER B SIP ITSP Server A : Branch Server ( with billing cdr ) : Billing module for auditing calls at branch level Server B : Billing Server ( with billing cdr ) : Official statements get generating from this server I am trying to match my billsecs on server A TO that of server B The problem seems to be that server A is counting the ringing as well, not just the ANSwered ( person talking ) If the phone rings for 25 secs and we speak for 30 secs, billsec is 55 secs and duration is 55 secs CDR values for deposition are correct if if the call has not been answered or busy. ie NO ANSWER or BUSY on Server A. Asterisk 13 on branch server any ideas on how to trouble shoot this problemthanks Zakir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.2.0 Video issues
I will rebuild my asterisk with the options enabled ONT_OPTIMIZE and BETTER_BACKTRACES Then I will create the traces and post them as per your recommendations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Thursday, March 12, 2015 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) toufic.khre...@gmail.com wrote: Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running asterisk 1.6.2.6 I receive the following warning: [2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples for alawtolin core show channels Channel Location State Application(Data) IAX2/Mypbx1-15288(None) Up AppDial((Outgoing Line)) SIP/6000-000f(None) Up Dial(IAX2/Mypbx1/300,300,Tt) 2 active channels Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and GSM codecs) Voice is not very clear and choppy If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 , voice is very clear. Both Asterisk 1.6.2 and Asterisk 12 no longer receive bug fixes, so I'm going to skip past this issue. 2. Asterisk 13.2.0 Video issues (no IAX2 voice issues). Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175 (asterisk engine stops/crashes) Asterisk crashing is a bug. That's a bad thing. Please get a backtrace [1] and file an issue on the issue tracker [2]. A pcap of the message traffic would also be very helpful. Call from Groundwire video sip (IOS since Android version does not H264 codec) to Grandstream GXV3175, Asterisk stops I'm going to assume Asterisk stops means it crashed as well. If you'd like to get a backtrace for that as well and attach it to the same issue, that would be helpful - it may be the same problem that you see with the Bria phone, or it may be something else. Calls between SIP Video softphones works well no issues. Well, that's good. :-) Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well. (Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) Calls between GXV3275 and GXV3175 video streaming is very slow on the GXV3175 (this is not the case under Asterisk 12.8.1) Calls from GXV3175 to Bria (video is displayed on bria side only) Since there are some that work fine, and some that don't, the trick is going to be knowing: (1) How the SIP peers (or PJSIP endpoints) are configured (2) How the phones are negotiating media with Asterisk Both your SIP configuration as well as a DEBUG log - generated with trace logging, showing the negotiation [3] - will be needed to figure out what is occurring. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira/ [3] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP sent to internal IP
Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the public IP address as it is seen on the packet header. Signalling is flowing correctly with no issues. Could you please advise why is this happening and how to correct this? Here is the [peer] in my sip.conf and the SDP in the setup (INVITE + OK). I'll be happy to provide any other information if needed: Sip.conf: [peer_name] deny=0.0.0.0/0 permit=remote_public_IP type=peer host=remote_public_IP ; same as permit defaultip=remote_public_IP ; same as permit qualify=no nat=yes disallow=all allow=alaw context=CALL_in dtmfmode=rfc2833 codecprobe=yes canreinvite=yes video=no restrictcid=no insecure=invite trustrpid = yes The SDP from both the INVITE and OK packets (from TShark). 172.24.100.2 is the local-private IP address of the remote UA and 192.168.1.200 is the local-private IP of my Asterisk. Both public IP's are static and do not change: *INVITE* Internet Protocol Version 4, Src: 192.168.1.200 (192.168.1.200), Dst: remote_public_IP (remote_public_IP) User Datagram Protocol, Src Port: 65060 (65060), Dst Port: 5060 (5060) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:858@remote_public_IP SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 614275163 614275163 IN IP4 my_public_IP Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 my_public_IP Time Description, active time (t): 0 0 Media Description, name and address (m): audio 18468 RTP/AVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ptime:20 Media Attribute (a): sendrecv *200 OK* Internet Protocol Version 4, Src: remote_public_IP (remote_public_IP), Dst: 192.168.1.200 (192.168.1.200) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 65060 (65060) Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): default 1426152411 1426152411 IN IP4 172.24.100.2 Session Name (s): Quantum PBX V 3.0 1.8.17.1 cert Connection Information (c): IN IP4 172.24.100.2 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 32000 RTP/AVP 8 101 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): ptime:20 Media Attribute (a): maxptime:90 Thank you, Harel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
