Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

the monolithic dialplan applications have specific options that place channels into dialplan contexts that execute after their execution. I'm not even sure I can begin to wrap my head around what that will do to a channel in ARI. -- *Matthew Jordan* Digium - A Sangoma Company | Senior Vice President

Re: [asterisk-users] Community forum ?

_ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.aste

Re: [asterisk-users] getting invites to rtp ports ??

Aug 2018 22:52:05 -0400, > Matthew Jordan wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group > wrote: > > > > Depending on log trolling (Asterisk security log) misses a lot, and > also depends on the

Re: [asterisk-users] getting invites to rtp ports ??

gt; 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://l

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

00:21] -- AGI Script agi://127.0.0.1/route > completed, returning -1 > [Jul 20 20:00:21] == MixMonitor close filestream (mixed) > [Jul 20 20:00:21] == End MixMonitor Recording PJSIP/local-004e > > Nothing shows up in nc. > > P.S. I have no idea why it thinks the other prompts are

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

Fri, Jul 20, 2018, 11:45 AM Matthew Jordan <mailto:mjor...@digium.com>> wrote: > >> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim > <mailto:naftoli...@gmail.com>> wrote: >> >> Crickets... >> >> I've tried this now on 15.5.0. Stil

Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim wrote: > > Crickets... > > I've tried this now on 15.5.0. Still completely broken. > > I suspect you’re encountering behavior that is working as intended. Normally, when Asterisk plays back a file, it scans the file system for all files

Re: [asterisk-users] Comparison of PJSIP and SIP in Asterisk database

mostly involve shenanigans and/or custom code - than a second instance of Asterisk will understand and read that JSON just fine. Assuming it was told to get that information from its AstDB via Sorcery as well. -- Matthew Jordan

Re: [asterisk-users] PJSIP_AOR Slow

) > }"!=""]?Set(WORKINGPEERFOUND=1)) > > exten => example_839,23,NoOp(${PJSIP_AOR(example_240,contact)}) > > exten => example_839,24,ExecIf($["${PJSIP_AOR(example_240,contact) > }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_240)) > > exten

Re: [asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

nity 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.dig

Re: [asterisk-users] RTCP + Stasis causing high memory consumption

en those messages are generated. If that doesn't fix it, then you may have some form of malformed RTCP packet that is causing Asterisk to think that it has a slew of SR/RR reports to generate. You may want to look at the RTCP information in wireshark to determine how many RR/SR reports are be

Re: [asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

nitor or MixMonitor? With what application arguments? If you look at a packet capture, does the packet capture reveal anything about the jitter, and on what call leg? Have you tried using a JITTERBUFFER? -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35

Re: [asterisk-users] user-agent access from pjsip

can get that off of the Contact. You can get the Contact via AMI by listening for events and by querying for the status of the contacts [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PJSIPShowRegistrationInboundContactStatuses -- Matthew Jordan Digium, Inc. | CTO 445 Ja

Re: [asterisk-users] Asterisk 14 audio quality with remote files

_channel_name(chan), filename, ast_format_get_name(ast_channel_writeformat(chan)), preflang ? preflang : "default"); Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _

Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

requests in Kamailio. >> > > This seams easier, for the moment. > > I think I still need to better understand what are mixed Asterisk-Kamailio > architectures main strengths > compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that > is another stor

Re: [asterisk-users] asterisk13+app_queue scalability

've run into a situation where app_queue no longer scales for you, you need to build your own queuing solution using Asterisk's APIs. app_queue was not designed to scale across multiple Asterisk instances, nor was it designed to scale up infinitely (which, of course, nothing is.) Matt -- Matthew Jor

Re: [asterisk-users] packet loss stats - how does asterisk know about packets sent % lost ?

m the endpoint(s) in question. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote: > On 21-11-16 15:17, Matthew Jordan wrote: > > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens <jonas.kell...@telenet.be> > wrote: > >> Hello >> >> when using As

Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

> > I did not see this behaviour in previous Asterisk versions. > > Could this be a bug ? > > There's not enough information here to know what is preventing the call from occurring. I'd look at a debug log between the calle

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

ahdi kernel module to be installed and available. (I could be wrong on the need for a physical card however, so your mileage may vary.) - Upgrade to a version of the kernel that res_timing_timerfd supports. That should be Linux 2.6.26 and glibc 2.8 or later. Personally, if I were in your shoes, I'd go with t

Re: [asterisk-users] Asterisk 11.24.1 garbled audio

RTP traffic (along with potential jitter) - CPU utilization with an active conference (95% idle doesn't mean that some core isn't pegged) - Any potential transcoding issues or codec issues Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 -

Re: [asterisk-users] SIP and RTP port and IP addresses

-- 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

Re: [asterisk-users] Conference Call like conference done by mobile!

nd the new callee in the same bridge as the original callee. This process could be repeated as many times as you want. [1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_HOLD_INTERCEPT -- Matthew Jordan Dig

Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

nt name (login name), you can use the AMI_CLIENT [1] dialplan function to retrieve the number of sessions they have currently established. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_AMI_CLIENT -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

on state or by releasing them back to the dialplan. -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and

Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

an by "stopped responding"? > Alternatively, how can dialplan check if there is any AMI user connected and > decide dial plan execution? > > Thanks & Regards, > Amit Patkar -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Ch

Re: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?

ide of Asterisk's control (via attended transfers). Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidt

Re: [asterisk-users] Dial and start music on hold after timeout

an play media back yourself using MoH or one of the other sound generation applications. (3) Wait for one of your outbound channels to pass a 180 back, and allow that to cause the inbound channel to ring. [1] http://lists.digium.com/pipermail/asterisk-users/2016-August/289781.html -- Matthew J

Re: [asterisk-users] SIP 603 response when call is not answered

vicestate.c:467 in do_state_change: Changing state for > SIP/111 - state 1 (Not in use) > 51 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state > '1' > While it's a bit harsh, there's nothing inherently wrong with returning a 603 in this case - so I wouldn't say it's

Re: [asterisk-users] Leave and re-enter a conference

call in another participant. Note that you need to use Originate instead of Dial, as you would otherwise have the participant be bridged in a new bridge with whoever they dialed. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge -- Matthew Jordan Digium,

Re: [asterisk-users] PJSIP not detected

e found on the wiki here: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-Troubleshooting -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Hunts

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

upported. Supported timelines for versions are available on the wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://di

Re: [asterisk-users] Asterisk & Vitelity Invite issues

ng sent to them, there's something seriously wrong with that provider. This is pretty core functionality in any SIP stack. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.c

Re: [asterisk-users] Asterisk 13 High CPU usage

spinning. Can you get a backtrace of the threads? [1] Make sure you have DONT_OPTIMIZE and BETTER_BACKTRACES enabled. That should show us what the threads are doing, which would give us a better idea of what is spending all the time

Re: [asterisk-users] Asterisk 13.10.0 Now Available

ly to improve Asterisk's support of available ciphers both in DTLS and SRTP. Thanks Alexander! -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _

Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

/goautodialce/wiki/goautodial_getting_started_guide As a result, you will almost certainly need to solicit help from the GOautodial folks. Things that are packaged up in such a fashion typically have a specialized configuration that is too specific for the Asterisk project itself to support. Matt -- Mat

Re: [asterisk-users] PJSIP and the pound (#) as %23

#" is excluded > because it is used to delimit a URI from a fragment identifier in URI > references (Section 4). The percent character "%" is excluded because > it is used for the encoding of escaped characters. > > delims = "<" | ">" | "

Re: [asterisk-users] Function SHELL not registered

zing data passed through to said dialplan functions, and should implement their own stringent access control. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.c

Re: [asterisk-users] VoipRaider is true for FREE calls?

the asterisk-biz list: http://lists.digium.com/mailman/listinfo/asterisk-biz While I know conversations tend to diverge sometimes, the asterisk-users list should be about using Asterisk, and not about promoting some third party service or software that may pertain to Asterisk. -- Matthew Jordan Dig

Re: [asterisk-users] Dial command for SIP driver with To-header config

', that'd probably be categorized as an improvement to this option. -- 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 -- __

Re: [asterisk-users] CDR ODBC error

ially a replacement for it. cdr_odbc doesn't receive much attention as a result. Frankly, we should probably just remove cdr_odbc. -- 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

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

quot;value": "yes"}, {"attribute": "rtp_symmetric", "value": "yes"}, {"attribute": "context", "value": "default" }, {"attribute":

Re: [asterisk-users] PJSIP Returning 421 Extension Required

rc1 > Content-Length: 0 > > PJSIP is rejecting the inbound INVITE request as 100rel is required, but is not in the Supported header of the inbound SIP INVITE request. I would suspect that the UAC is doing things incorrectly by placing 100r

Re: [asterisk-users] DIALSTATUS not being set

ult, 1000, 2) exited non-zero on 'PJSIP/bob-0000' -- Executing [h@default:1] NoOp("PJSIP/bob-", "") in new stack -- Executing [h@default:2] Log("PJSIP/bob-", "NOTICE, ANSWER") in new stack [Dec 22 16:34:17] NOTICE[9740][C-000

Re: [asterisk-users] Asterisk CLI and database problem

safe_asterisk script under a user without sufficient permissions, and/or running/invoking the Asterisk CLI (via "asterisk -rv") as a user with insufficient permissions. I would double check: (1) What user/groups own the various Asterisk directories (specified in your asterisk.conf)

Re: [asterisk-users] asterisk 13 n-way call problem

1/6000-436", > "DYNAMIC_FEATURES=") in new stack > -- Executing [5082@dynamic-nway:6] MeetMe("DAHDI/i1/6000-436", > "5082,1pdMXq") in new stack > == Spawn extension (sipphonesconf, 6000, 4) exited non-zero on > 'SIP/5082-0046' > > > Is this expected

Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

C driver or the Maria database. > > If you don't see that message, then something is preventing those events > from getting delivered inside of Asterisk, which would only occur if you > had some other serious call related issue

Re: [asterisk-users] CEL entries over ODBC several hours late

't show up in the database, then it is either in the ODBC driver or the Maria database. If you don't see that message, then something is preventing those events from getting delivered inside of Asterisk, which would only occur if you had some other serious call related issues occurring

Re: [asterisk-users] host parameter equivalent in pjsip.conf

-0975-4ff0-bd3b-bd5c38e594c4 > > To: <sip:+12345...@test.com> > > Contact: <sip:+12345678@4.3.2.1:60938> > > Route: > > User-Agent: Asterisk PBX 13.6.0 > > In order to preserve the request URI, you'll need to s

Re: [asterisk-users] after upgrade buttons on Dahdi phones don't work [SOLVED]

x was to change "from-internal" to > "internal" in dahdi-channels.conf . So that just leaves the question of how > this configuration ever worked at all. > > Sounds like you may have hit step 6... http://plasmasturm.org/log/6debug/ -- Matthew Jordan Di

Re: [asterisk-users] endwhile jumping out of macro

hannel is executing within them. Terminating an outer container of PBX flow without properly terminating an inner one can inbalance the stack. And just as a reminder, Macros are deprecated. They tend to have odd side effects at times, and overly nesting Macros can result in a crash. You should

Re: [asterisk-users] endwhile jumping out of macro

On Mon, Nov 30, 2015 at 11:34 AM, Ethy H. Brito <ethy.br...@inexo.com.br> wrote: > On Mon, 30 Nov 2015 09:40:50 -0600 > Matthew Jordan <mjor...@digium.com> wrote: > > > On Sat, Nov 28, 2015 at 7:14 AM, Ethy H. Brito <ethy.br...@inexo.com.br> > > wrote: &g

Re: [asterisk-users] PJSIP and RTT in realtime

why it generally does not show up in configuration documentation. However, since this is a sorcery object, you can specify in sorcery.conf where you'd like that object to be persisted. Note that by default, it is persisted using the 'memory' wizard. -- Matthew Jordan Digium, Inc. | Director of

Re: [asterisk-users] pjsip show xxxx like endpoint?

On Sun, Oct 18, 2015 at 12:39 PM, George Joseph <george.jos...@fairview5.com> wrote: > Did you open a Jira issue for this yet? I can actually work on this this > week. > I think it'd be pretty cool. George: want me to open an issue? -- Matthew Jordan Digium, Inc. | Director of

Re: [asterisk-users] Sending XML over the asterisk PJSIP

You can send arbitrary text message to/from Asterisk using SIP MESSAGE requests. The fact that the text is XML would be immaterial to Asterisk. That's probably the closest way to send arbitrary data to Asterisk without writing a specific new module in the PJSIP stack. -- Matthew Jordan Digium, Inc. |

Re: [asterisk-users] Tim's band DEEPFALL NOT SPAM!!

; > Please share this link with anyone you might know that could spare $5 toward > a good cause. > I'm pretty sure this has nothing to do with the Asterisk project. Please don't e-mail this list again with non-Asterisk related questions or topics. Doing so will get you kicked off

Re: [asterisk-users] Sending XML over the asterisk PJSIP

On Sat, Oct 17, 2015 at 2:32 AM, Thyda ENG <ength...@gmail.com> wrote: > Can i send XML data over the asterisk PJSIP ? > That's a fairly generic question. Can you be more specific about what you are trying to accomplish? -- Matthew Jordan Digium, Inc. | Director of Technology 4

Re: [asterisk-users] pjsip database error when using MS SQL via ODBC

y_timeout' and 'qualify_frequency'. Which one is currently giving the conversion error? Matt 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 -- _

Re: [asterisk-users] Help with voicemail

navail.gsm... > > Can someone help me to solve my problem? > Do you have a g729 codec module loaded? If so, does it show a translation path between g729 and gsm? If so, do you have sufficient encoder/decoder licenses? Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Ja

Re: [asterisk-users] Semicolon use in configuration?

pjsip’s Contact User to > do it by specifying the User portion. > > However semi-colon is treated as a comment by the Asterisk parser. Adding > quotes (“) around the setting doesn’t seem to help. > Use a '\', i.e., contact=sip:01234567\;tgrp=01234567\;trunkcontext=... -

Re: [asterisk-users] Storing HANGUPCAUSE in CDR

On Fri, Oct 9, 2015 at 8:27 AM, Ross Beer wrote: > Hi Andrew, > > Unfortunately that has stopped working when using chan_pjsip and asterisk > 13. > > The CDR is closed too early after a dial attempt. This is the expected > behaviour for Asterisk 13, however you should be

Re: [asterisk-users] PJSIP: how to retrieve underlying SIP Call-ID

hould be in the latest RC (13.6.0-rc2 [2]). In either case, you're using a function as opposed to some application, which means you do need to call the functions on the specific channel. To get access to the outbound channel, you can use a pre-dial handler's 'b' option [3]. The Call-ID *should* be set up on

Re: [asterisk-users] does res_pjsip support ZRTP?

your use cases. Would I like it to work well for you? Of course! But if you don't participate by reporting issues, testing changes, and contributing code, there's not much I can do for you other than to note that the line is long, and feel free to stand in it until someone in t

Re: [asterisk-users] Respond to an out of call SIP MESSAGE

On Mon, Sep 28, 2015 at 11:38 AM, Emil Ohlsson wrote: > Ah, so I can use > > MessageSend(sip:alice) > > to send a message to Alice then (reusing the existing TLS session). That does > seem to work. Thanks :-). I didn't know you could use users there. > > Is there a variable or some

Re: [asterisk-users] Respond to an out of call SIP MESSAGE

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend SendText is used for sending text messages within a call. Since a SIP channel is not servicing the out of call text message, you cannot use it to send a SIP MESSAGE request back to whatever sent the original SIP MESSAGE reque

Re: [asterisk-users] res_pjsip. Turn off the authorization request for an incoming MESSAGE

know that requests for that endpoint should not be authenticated, then you can remove the auth option from the endpoint and it should allow the request to proceed without a 401 challenge response. If you need to authenticate certain requests while allowing others through, then today, there is no wa

Re: [asterisk-users] Problem with Cisco CUBE when dialling into Asterisk 13 server

o send Asterisk a BYE, there's not much anyone can tell you unless they are familiar with that device. Asterisk is being told to hang up the call, and so it will do so. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW -

Re: [asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13

list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)

. The upgrade was painless, since we stayed in the 1.8 range, we did not have to modify any of our config files or dialplans. Maybe this can assist someone else struggling with older 1.8 series timer issues. Regards Always nice to hear that we fixed things. Thanks for the follow-up! -- Matthew

Re: [asterisk-users] Siren7 for Asterisk 13.5

(I forget why). Alas, until we get off our butts, yes. Sorry about that. Really, we're putting as much effort into fixing things and issues that affect a lot of people. While siren7/siren14/silk are nice, there aren't as many people using them as other affected things at this moment. -- Matthew

Re: [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13

://lists.digium.com/mailman/listinfo/asterisk-users -- 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

Re: [asterisk-users] Asterisk 11.19.0 Now Available

, not 11.18.0 How can we apply this patch to a legacy asterisk-11.18.0 tar.gz ? Thanks for any hint. That's a bug in the release scripts, which had to be rewritten when we moved to Git. We'll try to get it sorted out for the next release. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan

Re: [asterisk-users] Filters

present in some calls I push through my asterisk. Thanks If you're willing to write C, then yes, what you're looking to do is possible. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] Messages out of calls. Is it really possible?

/Asterisk+13+Function_MESSAGE and https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE_DATA [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] Can't install gmime22

, you really don't need that dependency. -- 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

Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels

channels. That fact that you have two different SIP channels means that something either performed two Originates, or you have done a parallel Dial. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com

Re: [asterisk-users] Distributed Device States - Best Option

-- 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

Re: [asterisk-users] error trying to get PJSIP working

for the realtime tables for PJSIP has been updated many times, as new features have been added. The alembic scripts bundled with Asterisk can manage your DB schemas for you, or can be used to generate the schema used by your specific version of Asterisk. -- Matthew Jordan Digium, Inc. | Director

Re: [asterisk-users] Calling multiple phones at ones

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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

Re: [asterisk-users] queue periodic-announce without stopping ringing

information on ARI and its intended use, see [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-play [2] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan

Re: [asterisk-users] ARI echo test

tutorial example should give you an ARI event over a WebSocket connection. https://wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] DPMA - Asterisk Realtime

of it, and we'll keep evaluating it versus other planned and requested features. Thanks - Matt -- 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

Re: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock

backtrace. Instructions for both can be found here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- 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

Re: [asterisk-users] Asterisk proxying a REFER

information about the channels on the PBX/Adhearsion server, who sends the REFER request, and what happens explicitly in the scenario? Matt -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] Asterisk 11 SRTP: unsupported crypto parameters: UNENCRYPTED_SRTCP

to not send the attribute. -- 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

Re: [asterisk-users] meetme vs confbridge max user comparison wanted

MeetMe hit a limit at around 60 channels, and ConfBridge reach over 240 channels. Worst case for ConfBridge was around 140 channels. Note that the ConfBridge sample rate, mixing interval, and other parameters can greatly affect how far it scales out. -- Matthew Jordan Digium, Inc. | Director

[asterisk-users] Asterisk is moving to Git

project! Matt [1] https://gerrit.asterisk.org [2] https://git.asterisk.org [3] http://lists.digium.com/mailman/listinfo/asterisk-dev -- 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

Re: [asterisk-users] Asterisk 13.3.0 Centos Package Install Error

-2.3-5.el6.i686 (epel) Not found Does anyone have any idea what might be wrong? I just ran this on a CentOS 6.6 64-bit VM and couldn't reproduce the issue you're seeing. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] CDR dst value null after attended transfer

on the second line is missing. Am I doing something wrong here or this is a bug? Looks like you're hitting this bug: https://issues.asterisk.org/jira/browse/ASTERISK-24443 -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

on generating a backtrace can be found on the wiki here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- 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

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

that whatever you dialled exists. -- 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

Re: [asterisk-users] how asterisk detects silence?

-- 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

Re: [asterisk-users] PJSIP - Video Support for WebRTC

video on webrtc in asterisk 13 Please stop spamming the list with this e-mail. Resending it multiple times is clearly not yielding the results you'd like. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

is still flowing through Asterisk, it just isn't being decoded and passed through the core. -- 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

Re: [asterisk-users] Asterisk 13 : SILK codec ?

On Thu, Mar 19, 2015 at 1:22 PM, Steve Murphy m...@parsetree.com wrote: On Wed, Oct 29, 2014 at 7:10 PM, sean darcy seandar...@gmail.com wrote: On 10/29/2014 08:06 PM, Matthew Jordan wrote: On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote: Can we expect a SILK codec

Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

, Asterisk will tell you if the bridge is locally or remotely bridging the two channels. -- 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

Re: [asterisk-users] pjsip: outofcall_message_context

for. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_message_context -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

is bridging the two channels? -- 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

Re: [asterisk-users] Asterisk 13.2.0 Video issues

) Something is deleting the core files. (3) The core files are hiding really, really well. Either way, if you can't get a backtrace, there isn't much we can do to help with that problem. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Asterisk 13.2.0 Video issues

to find where the core file was located - this is typically determined by /proc/sys/kernel/core_pattern -- 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

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