[asterisk-users] Wanted: professional softphone
Hello! Does anybody by chance know of a softphone which additionally has a management suite to fully configure it userID based for Windows clients? Any idea is appreciated! Thanks Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delayed RTP start
Hi, I am debugging an issue that unfortunately involves two NAT instances - the firewall at our customer site, and the firewall in front of their Amazon instance. I have an HTEK phone at the customer site registering to the public address of the Amazon instance running asterisk (and FreePBX). This seems to work fine, and it can call local services (like fpbx *65 to read back the extension) with no problems. If it tries to make an outbound outside call, the remote phone (my cell for example) rings, I answer it, but there is no audio in either direction for nearly exactly 16 seconds, every time. Then audio starts in both directions without issue. I did a packet trace on the phone itself and see 16 seconds of outbound RTP with no inbound, then suddenly RTP in both directions until the call ends. I did a packet trace on the asterisk side and see the call setup, then sixteen seconds of nothing (??), then RTP starts in both directions. In the asterisk console I see this bit of interestingness: [2019-07-24 13:21:02] DEBUG[1890]: chan_sip.c:29923 __start_session_timer: Session timer started: 78 - 710779684e62266a77b047b31e4 261da@10.0.116.239:60060 1768000ms -- SIP/ast01-024b answered SIP/7222-024a [.snip.] [2019-07-24 13:21:02] DEBUG[17928][C-01f1]: bridge_native_rtp.c:660 native_rtp_bridge_compatible_check: Bridge '3bfbf253-d34f- 45e2-abc3-75e590d81739' can not use native RTP bridge as channel 'SIP/ast01-024b' has DTMF hooks [.snip.] [2019-07-24 13:21:18] DEBUG[18003][C-01f1]: res_rtp_asterisk.c:4179 ast_rtp_write: Ooh, format changed from none to ulaw [2019-07-24 13:21:18] DEBUG[18003][C-01f1]: res_rtp_asterisk.c:4019 rtp_raw_write: Starting RTCP transmission on RTP instan ce '0x7fe17426e7c8' So my main question is, what would cause a sixteen second delay before the codec could be decided? This is Asterisk 13.25.0 on the customer Amazon instance... the "ast01" peer is ours also - one of our external gateways, also running 13.25.0. Thanks, -- Jeff LaCoursiere StratusTalk, Inc. 703 496 4990 x108 815 546 6599 cell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
>>> I have updated the wiki. The script can be found within the >>> contrib/scripts/sip_to_pjsip subdirectory of an unpacked download of >>> Asterisk 13 and forward. Got it! Thanks, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
On Wed, Jul 24, 2019, at 10:09 AM, Doug Lytle wrote: > I'm currently reviewing the Digium wiki on migrating from chan_sip to > res_pjip and I'm trying to access the script that is provided to help > with conversion. > > https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip > > It would appear that said server hosting the script is no responding or > the link is no longer valid. In addition to what Malcolm stated, it can also be viewed on the Github mirror[1] but the complete contents of the directory are needed - not just the single .py [1] https://github.com/asterisk/asterisk/tree/master/contrib/scripts/sip_to_pjsip -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
Howdy, I have updated the wiki. The script can be found within the contrib/scripts/sip_to_pjsip subdirectory of an unpacked download of Asterisk 13 and forward. Cheers On Wed, Jul 24, 2019 at 8:10 AM Doug Lytle wrote: > I'm currently reviewing the Digium wiki on migrating from chan_sip to > res_pjip and I'm trying to access the script that is provided to help with > conversion. > > > https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip > > It would appear that said server hosting the script is no responding or > the link is no longer valid. > > Doug > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Malcolm Davenport Digium - a Sangoma company | Senior Product Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Tel/Fax: +1 256 428 6252 malco...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
Works for me from Comcast! John Novack Doug Lytle wrote: I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] svnview.digium.com down?
I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
