search for: softmix

...=================== ====== ================ ================ ================ ================ SIP/8484-00000000 Am conf-adminmenu 8484 SIP/8484-00000001 m conf-menu 8484 SIP/8484-00000002 Am conf-adminmenu 8484 -- Channel SIP/8484-00000001 left 'softmix' base-bridge <485fcffc-49ad-4e86-8d1b-4655631a232a> -- Channel CBAnn/1701414-00000000;2 joined 'softmix' base-bridge <485fcffc-49ad-4e86-8d1b-4655631a232a> -- <CBAnn/1701414-00000000;1> Playing 'confbridge-leave.gsm' (language '') -- Channel...

Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of type softmix even if two channels only are involved). I can do this with ARI of course, but handling transfers will become quite difficult : I will receive r...

...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...

After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter and then exit a conference room, I see: -- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> -- Channel CBAnn/207-0000067f;2 left 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> I'd expect those channel to immediately go away, but they just stay around: asterisk*CLI> core show channel CBAn...

...551 at extensions:5] Set("SIP/199.73.108.66-00000003", "CONFBRIDGE(user,template)=church") in new stack ??? -- Executing [2551 at extensions:6] ConfBridge("SIP/199.73.108.66-00000003", "9999") in new stack ??? -- Channel SIP/199.73.108.66-00000003 joined 'softmix' base-bridge <6dfaada2-2d01-44e4-b025-914ca18eee25> Bridge Enter message for channel SIP/199.73.108.66-00000003: 1425186138.00529497 0x7fbbe8001628 - Updating Party A SIP/199.73.108.66-00000003 snapshot 0x7fbbe8001628 - Processing bridge enter for SIP/199.73.108.66-00000003 0x7fbbe8001628...

...; Playing 'conf-onlyperson.g722' (language 'de') -- <PJSIP/7000-00000000> Playing 'confbridge-join.g722' (language 'de') -- <CBAnn/8600-00000000;1> Playing 'confbridge-join.slin' (language 'en') -- Channel CBAnn/8600-00000000;2 joined 'softmix' base-bridge <52997aa1-eb00-481c-8c56-e26d78d01515> -- Channel CBAnn/8600-00000000;2 left 'softmix' base-bridge <52997aa1-eb00-481c-8c56-e26d78d01515> -- Started music on hold, class 'default', on channel 'PJSIP/7000-00000000' -- Channel PJSIP/7000-00000000...

...ification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] => ConfbridgeMute [Privilege] => call,all [Conference] => 42 [BridgeUniqueid] => 9f2ae5df-0749-4494-b8b7-12eb50dc765d [BridgeType] => base [BridgeTechnology] => softmix [BridgeNumChannels] => 2 [Channel] => SIP/5000-00000006 [ChannelState] => 6 [ChannelStateDesc] => Up *[CallerIDNum] => 5000* [CallerIDName] => 5000 [ConnectedLineNum] => <unknown> [ConnectedLineName] => <unknown> [AccountCode] => [Context] => from-sip [Exte...

...pgraded 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...

..., 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third channel comes in. I thought I could use ConfBridge for that, but it creates a bridge that is not smart (it is of type softmix even if two channels only are involved). I can do this with ARI of course, but handling transfers will become quite difficult : I will receive replac...

...is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881] VERBOSE[6921] bridge_channel.c: Channel CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f joined 'softmix' base-bridge <23340bca-6823-4c70-a395-e3b092aeb671>...

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...

Folks, I've read some sources claiming that Asterisk does not need DAHDI for timing in 1.6.1. Is this true? Searching the web, all I can find is pages celebrating the fact but no actual documentation on which version it was introduced in and how one would go about configuring an external time source. I'm having a devil of a job trying to compile DAHDI on a hosted Xen VM and thought I

...IDGE(bridge,record_file)=/var/spool/asterisk/confbridge/296955.wav)") in new stack -- Executing [s at TTMConferenceTalker:8] ConfBridge("SIP/sbc1-00002426", "296955,default_bridge,TTM_profile,TTM_profile_menu") in new stack -- Channel SIP/sbc1-00002426 joined 'softmix' base-bridge <f14cbf31-d82e-4e2f-9c62-3f70b2c66165> Here is a sample log from asterisk 11 [Oct 27 15:08:48] VERBOSE[14718][C-0005278d] pbx.c: -- Executing [s at TTMConferenceCaller:5] ExecIf("SIP/sbc1-0004e882", "1?SET(CONFBRIDGE(bridge,record_conference)=yes)&quo...

...:4875 pbx_extension_helper: Launching 'ParkAndAnnounce' -- Executing [s at macro-parkswitch:1] ParkAndAnnounce("SIP/vpn-sbc-00000001", ",,PARKED,SIP/100") in new stack [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 find_best_technology: Bridge technology softmix does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 find_best_technology: Bridge technology simple_bridge does not have any capabilities we want. [Jan 30 21:00:01] DEBUG[7118][C-00000000]: bridge.c:486 find_best_technology: Bridge technology native_rt...

...forbidden while getting details [2017-06-15 07:43:57] DEBUG[25198][C-00000004]: bridge.c:506 find_best_technology: Bridge technology native_rtp is not compatible with properties of existing bridge. [2017-06-15 07:43:57] DEBUG[25198][C-00000004]: bridge.c:496 find_best_technology: Bridge technology softmix does not have any capabilities we want. [2017-06-15 07:43:57] DEBUG[25198][C-00000004]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip/pjsip_distributor.c:785 distribute: rdata clone remove di...

...by Ross Beer) * ASTERISK-28608 - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson) * ASTERISK-28618 - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) * ASTERISK-26481 - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) * ASTERISK-28572 - Memory leaks in res_calendar_exchange and...

...s_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson) * ASTERISK-26481 - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) * ASTERISK-28618 - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) * ASTERISK-28616 - parking: Deadlock when multi call parking (Reported by Joshua C. Colp) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28572 - Memory l...

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

...tam Khankishyiev) * ASTERISK-24312 - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) * ASTERISK-24426 - CDR Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants (Reported by Matt Jordan) * ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sendi...