search for: bridge_softmix

...=1&t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simp...

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-us...

...ently upgraded from Asterisk 11 to Asterisk 13. > > Most of it went smoothly thanks to the documentation detailing how to > upgrade to 12 and then how to upgrade to 13. > > The only thing that didn't work correctly was Music On Hold. Eventually > I tracked this down to using bridge_softmix instead of bridge_simple. > > What I'm asking is, does anyone have any explanation as to why MOH would > not work with bridge_softmix? Asterisk 11 had been working for at least a > year with bridge_softmix and the MOH was fine. With the same configuration > (almost) Asterisk 13...

...fsoftware.co.uk> wrote: > Thanks Richard. This is exactly the answer I was looking for. > > > I'm now assuming that Asterisk 11 was using it's equivalent > "bridge_simple" but I was getting confused because the only bridge module I > saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that > would have been the only bridge getting loaded at first. > > > Is it expected that if bridge_softmix handled a normal two party call > then MOH would no longer function? > That is correct. bridge_softmix is optimized for multi-party confere...

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

...Joshua Colp) * ASTERISK-27553 - [patch] res_curl: Avoid error message on unload. (Reported by Alexander Traud) * ASTERISK-27557 - [patch] clang 5.0: implicit conversion to char changes value to negative. (Reported by Alexander Traud) * ASTERISK-27550 - [patch] bridge_softmix: Avoid warning about an uninitialized variable. (Reported by Alexander Traud) * ASTERISK-27559 - [patch] editline: Avoid comparison between pointer and zero character constant. (Reported by Alexander Traud) * ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negat...

...gcc OSX with 64 bit integers (Reported by Corey Farrell) * ASTERISK-24796 - Codecs and bucket schema's prevent module unload (Reported by Corey Farrell) * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML (Reported by Ashley Sanders) * ASTERISK-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill (Reported by JoshE) * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime fail (Reported by Terry Wilson) * A...

...writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERIS...

...a C. Colp) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cer...

...writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERIS...

...a C. Colp) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-16.8-cer...

...d by Jared Hull) * ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled (Reported by Torrey Searle) * ASTERISK-26686 - res_pjsip: Lock inversion in transport management (Reported by Ross Beer) * ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11. (Reported by Alexander Traud) * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE (Reported by Joshua Elson) * ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shu...

...To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim...

...To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim...

...To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim...

...To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim...

...ngtae kim) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any paramet...

...ngtae kim) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any paramet...

...by Joshua C. Colp) * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any paramet...

...by Joshua C. Colp) * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any paramet...