Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen ATA
George Thank you we will get 13.18.4 on the test box ASAP, and thank you for the wiki link so we can pull together additional information to resolve this issue. I have upgraded to 13.18.4 and the system is not crashing now. Is there any specific item that may point to why the issue was occurring? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "George Joseph" Sent: Tuesday, December 19, 2017 2:59 PM To: brya...@zktech.com, "Asterisk Developers Mailing List" Subject: Re: [asterisk-dev] Asterisk 13.17.1 Crash on ConfBridge - NetGen ATA On Tue, Dec 19, 2017 at 12:45 PM, Bryant Zimmerman wrote: We are having an issue with asterisk 13.17.1 dumping when a call from a NetGen Smart ATA drops into a confbridge The call props up and then things just go wrong. I have talked with the support guys at NetGen and they have requested I work start with the asterisk dev group so we can figure out what is causing this issue and why asterisk is dumping. They are willing to fix anything from their end but we have not been able to figure out what in their rtp stream is triggering this. Their ATA's seem to work out side of the confbridge without issues so far. Any ideas are appreciated. The asterisk dump is by far my biggest concern. Below is the first part of the dump Backtrack. I have attached a copy of the complete Backtrack. I need to know what more would be needed to get to the bottom of this issue. As it stands now the NetGen Smart ATA will cause asterisk 13 to crash if placed into a confbridge. http://www.netgencommunications.com/ The support guy said we could contact them at supp...@netgencommunications.com There's not a whole lot of info in this backtrace for us to really know what's going on but can you try with 13.18.4? There have been recent crash fixes that may help. If 13.18.4 doesn't help, recompiling with debugging turned on, re-creating the issue, then following the wiki instructions to get a backtrace will help us figure out what's up. https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Executing [s@Core_ConfBridge_Basic:11] ConfBridge("PJSIP/6162480909.351-", "6162480909.~Promo~GA1,,,sample_user_menu") in new stack > 0x7f3ff800b4c0 -- Probation passed - setting RTP source address to 192.168.209.194:10020 -- Channel CBAnn/6162480909.~Promo~GA1-;2 joined 'softmix' base-bridge <4a00cdad-91cb-4924-8abe-8dc9cad08f10> -- Playing 'conf-onlyperson.ulaw' (language 'en') UBNTU-ROSSI-GUEST*CLI> *** Error in `/usr/sbin/asterisk': malloc(): memory corruption: 0x7f3fac00c220 *** === Backtrace: = /lib/x86_64-linux-gnu/libc.so.6(+0x777e5)[0x7f402147b7e5] /lib/x86_64-linux-gnu/libc.so.6(+0x8213e)[0x7f402148613e] /lib/x86_64-linux-gnu/libc.so.6(__libc_malloc+0x54)[0x7f4021488184] /usr/sbin/asterisk(ast_json_malloc+0xa)[0x52a23a] /usr/lib/x86_64-linux-gnu/libjansson.so.4(json_object+0xb)[0x7f40227ab7bb] /usr/lib/x86_64-linux-gnu/libjansson.so.4(+0x6505)[0x7f40227aa505] /usr/lib/x86_64-linux-gnu/libjansson.so.4(json_vpack_ex+0x99)[0x7f40227aaa09 ] /usr/sbin/asterisk(ast_json_vpack+0x34)[0x52b6a4] /usr/sbin/asterisk(ast_json_pack+0xa1)[0x52b7c1] /usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x10df3)[0x7f3f90994df3] /usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x11a99)[0x7f3f90995a99] /usr/lib/asterisk/modules/res_rtp_asterisk.so(+0x13bcb)[0x7f3f90997bcb] /usr/sbin/asterisk(ast_rtp_instance_read+0x36)[0x588076] /usr/lib/asterisk/modules/chan_pjsip.so(+0x8cd7)[0x7f3f7c57ccd7] /usr/sbin/asterisk[0x4bc042] /usr/sbin/asterisk[0x50f3f1] /usr/sbin/asterisk(ast_stream_and_wait+0x56)[0x511bbe] /usr/lib/asterisk/modules/app_confbridge.so(+0xb716)[0x7f3f915d7716] /usr/lib/asterisk/modules/app_confbridge.so(+0xd5f6)[0x7f3f915d95f6] /usr/sbin/asterisk(pbx_exec+0xbd)[0x579155] /usr/sbin/asterisk[0x56e0c3] /usr/sbin/asterisk(ast_spawn_extension+0x18)[0x56feb8] /usr/lib/asterisk/modules/app_macro.so(+0x2c02)[0x7f3f6f8a6c02] /usr/sbin/asterisk(pbx_exec+0xbd)[0x579155] /usr/sbin/asterisk[0x56e0c3] /usr/sbin/asterisk[0x5703d1] /usr/sbin/asterisk[0x57190b] /usr/sbin/asterisk[0x5e45fd] /lib/x86_64-linux-gnu/libpthread.so.0(+0x76ba)[0x7f4021f436ba] /lib/x86_64-linux-gnu/libc.so.6(clone+0x6d)[0x7f402150b3dd] Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org --
Re: [asterisk-dev] One sip stack to rule them all....
I would agree with this. We have tried to deploy pjsip several times over the last year with limited success. We have had nothing but issues with database real-time deployments. Tables not working from one 13.x release to another. Table builders sorcery failing out. Issues when there are multiple transports on varying networks were udp is not routed correctly through the asterisk servers. No matter the settings. Connectivity issues with varying success by carrier. Unexplained audio quality issues that don't occur on the same spec running chan_sip We want to move to pjsip but the functionality and stability have only proven out for limited applications. Bryant From: "Daniel Journo" Sent: Sunday, October 8, 2017 3:12 PM To: "Asterisk Developers Mailing List" Subject: Re: [asterisk-dev] One sip stack to rule them all > What is _also_ needed, however, is more use of PJSIP and reports of specific > problems, and specific deficits of PJSIP so that the fear can be eased > before, at some point many years from now, chan_sip just doesn't work any > more. There are a number of specific issues on issue tracker which still need addressing before more people will take it on properly. Some issues probably require a semi-major rethink and probably won't be dealt with for months. Making chan_sip depreciated would leave Asterisk with no production grade sip stack that is officially being maintained. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Park patch to silence slot number
You can create dynamic lots with different contexts in the fly. You are actually parking at a context you just use an extension to identify access to the slot. If you have any identifying unique account info. Each lot is based on a parking lot context as a template. The lot name is dyanmically unique per lot. When you first create it. We use unified dialplan contexts for parking. We just augment our lots based on the account number for each client (prkAcctNumLotNum.SlotNym). It is very flexible you can pass in all the parameters on the fly using a combination of special channel dynamic variables and options in the park and parked command. We even specify dynamic ring back contexts with parameters in them so ring backs can be directed correctly. And accounts can only ever pickup their lots. We have customers with multiple parking lots all on the same account and the parking extensions are set in a database. The only limitation is you can't change the parking template context or shrink it once a lot is created without a restart of Asterisk. You can grow the number of parking slots dyanmically. The way we do this when you park directly on a slot the ring back time can be specified on a per slot basis as well. Customers love this. Sent from my Windows 10 phone From: David Cunningham Sent: Tuesday, October 3, 2017 6:46 PM To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] Park patch to silence slot number Hi Byrant, Thank you for the reply (and voicemail). Are you referring to the patch by Igor Goncharovsky? We had issues because it requires a different context for each overlapping parking slot, and our multi-tenant environment doesn't use a different context per tenant. What he said is below. However, we've just noticed that Asterisk 13 has an 's' option on the Park command to silence the slot number, so sorry for the trouble! "It is the how parking works in core of asterisk. When call parked, asterisk automatically put extensions for call park and for taking a call from park. I.e. when you want to park two calls at same extension and same context asterisk unable to do it, because unable to insert second time same extension in dialplan. I see no problem in using different extensions range for all parking lots in same context of using different contexts for all users" On 3 October 2017 at 16:23, Bryant Zimmerman wrote: > You don't need a patch this is possible with the current tools if you keep > track of parks pickups and ring backs. We use the current system to do > dynamic parks all of the time in multi tenant environment . We create > dynamic lots per tenant and address them per sub account. This allows for > each tenent to have parking slots with any number even when others use the > same. I paid part of the bounty to get the original dynamic parking system > working. > > > > Sent from my Windows 10 phone > > > > *From: *David Cunningham > *Sent: *Monday, October 2, 2017 11:09 PM > *To: *asterisk-dev@lists.digium.com > *Subject: *[asterisk-dev] Park patch to silence slot number > > > Hello, > > We'd like to get a patch written for Asterisk's Park command, so that with > a given option it won't play the parking slot number. The idea is that we > can allow multiple calls to be parked on the same apparent slot number by > playing the apparent slot number ourselves, and parking calls on a > different actual slot number. For example we'd play 701 but actually park > the call on 123701. > > Does anyone know of someone who'd be willing to write this patch, and > submit it for inclusion in Asterisk? We will of course pay a bounty. If I'm > asking in the wrong place then apologies and please let me know the right > one. > > Thanks in advance, > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 <+1%20213-221-1092> > Australia: +61 (0) 2 8063 9019 <+61%202%208063%209019> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev > -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Park patch to silence slot number
You don't need a patch this is possible with the current tools if you keep track of parks pickups and ring backs. We use the current system to do dynamic parks all of the time in multi tenant environment . We create dynamic lots per tenant and address them per sub account. This allows for each tenent to have parking slots with any number even when others use the same. I paid part of the bounty to get the original dynamic parking system working. Sent from my Windows 10 phone From: David Cunningham Sent: Monday, October 2, 2017 11:09 PM To: asterisk-dev@lists.digium.com Subject: [asterisk-dev] Park patch to silence slot number Hello, We'd like to get a patch written for Asterisk's Park command, so that with a given option it won't play the parking slot number. The idea is that we can allow multiple calls to be parked on the same apparent slot number by playing the apparent slot number ourselves, and parking calls on a different actual slot number. For example we'd play 701 but actually park the call on 123701. Does anyone know of someone who'd be willing to write this patch, and submit it for inclusion in Asterisk? We will of course pay a bounty. If I'm asking in the wrong place then apologies and please let me know the right one. Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.- Solved
I will just use sippeer to read the values I need. Thanks Bryant I am working on an application that uses sip messages to send sms. I am bumping into what I think may be a bug. The message is being sent to the asterisk server from the extension, but I have no access to the setvar variables or any variables that would normally be on a standard channel such as accountcode set for the peer for processing in the dial plan. Is there any way to access these outside of a call during message handling? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] sip Messages - No acces to variables set on the peer.
I am working on an application that uses sip messages to send sms. I am bumping into what I think may be a bug. The message is being sent to the asterisk server from the extension, but I have no access to the setvar variables or any variables that would normally be on a standard channel such as accountcode set for the peer for processing in the dial plan. Is there any way to access these outside of a call during message handling? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] ARI versioning in 13 and 14
+1 to option 2. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Issue AGI Get Full Variable - Solution
From: "Bryant Zimmerman" Sent: Tuesday, September 2, 2014 3:10 AM To: "Asterisk Developers Mailing List" Subject: Re: [asterisk-dev] Issue AGI Get Full Variable I am calling the GET FULL VARIABLE agi command. I am passing in the variable name and the channel name. It is responding with variable name back to the script and not the value. Failed AGI Rx << GET FULL VARIABLE aatAct0 SIP/6167761066.2012-0060 AGI Tx >> 200 result=1 (aatAct0) Expected AGI Rx << GET VARIABLE aatAct0 AGI Tx >> 200 result=1 (Macro~SBussniessMSIP-Operator~1) I am on asterisk 11.10.2 I am I using the Get Full Variable wrong any ideas? --- Ok I have figured this one out. The issue is the variable name must be wrapped in the ${variablename} format. The agi show commands topic get full variable docs do not show this format requirement. How do we get them updated so others do not get bit by this? -= Info about agi 'get full variable' =- [Syntax] get full variable [] Should read something Like: get full variable <${variablename}> [] Or Like: get full variable <${FUNC(variablename)}> [] [Description] Returns '0' if is not set or channel does not exist. Returns '1' if is set and returns the variable in parenthesis. This does an eval on all var expressions. should be included in ${variablename}, ${FNC(variablename)}, or $[expr] Understands complex variable names and builtin variables, unlike GET VARIABLE. Example return code: 200 result=1 (testvariable value) [Synopsis] Evaluates a channel expression [Runs Dead] Yes [See Also] Not available Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Issue AGI Get Full Variable
I am calling the GET FULL VARIABLE agi command. I am passing in the variable name and the channel name. It is responding with variable name back to the script and not the value. Failed AGI Rx << GET FULL VARIABLE aatAct0 SIP/6167761066.2012-0060 AGI Tx >> 200 result=1 (aatAct0) Expected AGI Rx << GET VARIABLE aatAct0 AGI Tx >> 200 result=1 (Macro~SBussniessMSIP-Operator~1) I am on asterisk 11.10.2 I am I using the Get Full Variable wrong any ideas? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk and video conferencing
Sangoma has trans coding solutions that allow for use with virtual machines. They had video codecs in their road map. I am not sure if they have them yet? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Dan Cropp" Sent: Monday, March 31, 2014 12:02 PM To: "Asterisk Developers Mailing List" Subject: Re: [asterisk-dev] Asterisk and video conferencing "I've been lobbying hardware manufacturers to provide video cards for Asterisk where we can have licenses to do transcoding and reformatting, so far with no success." I passed this onto someone in our hardware department to look into. I do worry about the thought of hardware for a video solution. We go into a lot of hospitals. They only want virtual servers. Not sure that a hardware based video solution will go over very well in many markets. For those worried about bandwidth of video, would it be possible to offload that work to another Asterisk box B? Put audio on box A. If you need video conferencing, have box A send that to box B? -Original Message- From: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Olle E. Johansson Sent: Monday, March 31, 2014 6:42 AM To: Asterisk Developers Mailing List Cc: Olle E Johansson Subject: Re: [asterisk-dev] Asterisk and video conferencing On 31 Mar 2014, at 12:47, Johan Wilfer wrote: > Hi! > > I've spent some time scratching my head thinking about video conferencing and how to go about it. Right now we use Meetme as a audio bridge for pstn connectivity and so on. But our users ask for video and screen sharing. I can see four distinct ways to go about this: > > 1. Asterisk right now - supports 1-1 video, and with confbridge 1 video stream can be sent to the other participants. (The codec must match thought, and a key-frame are not sent immediate if the video source is changed so there will be a garbled video stream until next key-frame.) > > 2. MCU - multiple video streams encoded in single stream. For video do the same as with audio. That means decode each stream, compose a new stream with all the participants layered out nicely. While this works for audio it consumes huge amounts of cpu to do this for video. > > 3. p2p - multiple video streams sent peer to peer. Each participant sends the audio/video to every other participant. This eats a lot of bandwidth for the users and can work for smaller conferences, but in a conference with 10 participants each will have to have a very good upstream connection. > > 4. Jitsi Videobridge - multiple video streams from server, but send only your stream to the server. The jitsi videobridge the distributes the stream to all other clients. This will eat a lot of bandwidth for the server, but not for the clients. This is also how Google Hangouts works. So if you are 10 participants you will send one stream to the server with your audio/video and receive 9 streams from the server for the other participants. > > > To be able to scale reasonably I think option 2 is out of the question. And option 3, p2p, eats to much bandwidth for the clients (and doesn't require an asterisk anyway). > > What you lose with option 4 is everything asterisk excels at: pstn connectivity, fine-grained control of each participant in the bridge. > > What are your thoughts on adding Jitsis approach in regards to video to Asterisk for confbridge or even ARI? No composing of video, just relaying the other participants streams to each other in the bridge. Then it's up to the client in the other end to display these streams in a reasonable way (like Google Hangout, and https://meet.jit.si/). Why? The jitsi video bridge exists and work fine :-) What you are forgetting here is the thing that has stopped us from doing really cool stuff with video - patents and licensing. The jitsi video bridge is a nice workaround, but not optimal if you have a lot of different devices. You put the load on the device and in bandwidth-constrained environments that's not good. Video is heavily dependend on peer2peer negotiation and doesn't really fit well in a PBX b2bua architecture... The jitsi model could work - but the SDP o/a handling would be really hard to get right in Asterisk. I've been lobbying hardware manufacturers to provide video cards for Asterisk where we can have licenses to do transcoding and reformatting, so far with no success. Cisco's H264 codecs recently became available for us in the Open Source world thanks to a generous solution by Cisco. I guess funding is needed to add anything cool to Asterisk using them. We can do MCU-style stuff, reformatting - but to do transcoding we need another codec :-) Google VP8 is around, I don't know what Digium's legal team have to say about us using it. Random