Re: [asterisk-users] Question on the RTP packet header
Hi Dan, Your best bet for looking at RTP media specifics is the standards that define RTP. Wikipedia has some really good resources on RTP and a list of the various RFC standards that relate: https://en.wikipedia.org/wiki/Real-time_Transport_Protocol On 8/28/23 11:16, Dan Cropp wrote: I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project. Using slin16 format. 1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is this correct? 2) Is there some place I can find a description of the 12-byte packet header fields? Dan -- _ -- 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] Question on the RTP packet header
I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project. Using slin16 format. 1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is this correct? 2) Is there some place I can find a description of the 12-byte packet header fields? Dan -- _ -- 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] Question about Sip Trunks who support Stir Shaken
> On 18 Aug 2023, at 04:50, Federico wrote: > > I am looking for a decent provider of SIP Trunks but it has to pass the Stir > Shaken token to the next carrier. Does anybody know about any? Sipstation > from Sangoma, does not support Stir Shaken. ( Case #01466843 / > 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) > > > I’d try Telnyx - where this works for me. And their online SIP debugging tool is second to none. Absolutely excels at finding issues with this sort of stuff. Dw. -- _ -- 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] Question about Sip Trunks who support Stir Shaken
Thanks. I have accounts with both companies and both have issues. From: asterisk-users On Behalf Of Dovid Bender Sent: Friday, August 18, 2023 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 382 and voicetel if you want an intro. On Thu, Aug 17, 2023 at 11:50 PM Federico mailto:feder...@digitalipvoice.com> > wrote: I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) Although it’s mandatory, somehow they think it’s ok. Go figure. -- _ -- 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 -- _ -- 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] Question about Sip Trunks who support Stir Shaken
Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 382 and voicetel if you want an intro. On Thu, Aug 17, 2023 at 11:50 PM Federico wrote: > I am looking for a decent provider of SIP Trunks but it has to pass the > Stir Shaken token to the next carrier. Does anybody know about any? > Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / > 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) > > Although it’s mandatory, somehow they think it’s ok. Go figure. > -- > _ > -- 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 > -- _ -- 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] Question about Sip Trunks who support Stir Shaken
Bandwidth.com, although there are minimums to meet. Cheers, Jeff LaCoursiere StratusTalk, Inc. On Fri, Aug 18, 2023 at 7:52 AM TTT wrote: > Check out Twilio > > > > > > *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On > Behalf Of *Federico > *Sent:* Thursday, August 17, 2023 11:49 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' < > asterisk-users@lists.digium.com> > *Subject:* [asterisk-users] Question about Sip Trunks who support Stir > Shaken > > > > I am looking for a decent provider of SIP Trunks but it has to pass the > Stir Shaken token to the next carrier. Does anybody know about any? > Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / > 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) > > Although it’s mandatory, somehow they think it’s ok. Go figure. > -- > _ > -- 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 > -- _ -- 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] Question about Sip Trunks who support Stir Shaken
Check out Twilio From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Sent: Thursday, August 17, 2023 11:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Question about Sip Trunks who support Stir Shaken I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) Although it's mandatory, somehow they think it's ok. Go figure. -- _ -- 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] Question about Sip Trunks who support Stir Shaken
I am looking for a decent provider of SIP Trunks but it has to pass the Stir Shaken token to the next carrier. Does anybody know about any? Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 / 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ]) Although it's mandatory, somehow they think it's ok. Go figure. -- _ -- 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] Question on ring count on incoming circuits
On 5/29/2023 4:12 PM, Steve Matzura wrote: On 5/28/2023 2:27 PM, Naveen Albert wrote: However, you can also pass audio without supervising (early media). You typically need to Progress() first to allow this, e.g. for SIP, or audio won't pass at all. ... If you want it to ring once and do something else, you could simply do: exten => s,1,Wait(6) ; 1 ring cycle is 6 seconds same => n,Answer(); answer, and do something else Just as you said at the top of this reply, no audio of any kind gets passed, so all the Wait(6) did was provide six seconds of dead-air silence before the outgoing message played. Oh well. Customers can't have everything. ;-) Well, yes, that's what you wanted, right? Or maybe I misunderstood. If you want people to hear *something* but not have it answer immediately, for those 6 seconds, amend that to: exten => s,1,Progress() same => n,Playback(foobar,noanswer) same => n,Answer() same => n,DoSomething() For example, this is common for playing an outgoing message or voicemail greeting, without supervising immediately, so if the caller hangs up before leaving a message, s/he is not charged for the call. Are you trying to do something like that? -- _ -- 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] Question on ring count on incoming circuits
On 5/28/2023 2:27 PM, Naveen Albert wrote: However, you can also pass audio without supervising (early media). You typically need to Progress() first to allow this, e.g. for SIP, or audio won't pass at all. ... If you want it to ring once and do something else, you could simply do: exten => s,1,Wait(6) ; 1 ring cycle is 6 seconds same => n,Answer(); answer, and do something else Just as you said at the top of this reply, no audio of any kind gets passed, so all the Wait(6) did was provide six seconds of dead-air silence before the outgoing message played. Oh well. Customers can't have everything. ;-) -- _ -- 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] Question on ring count on incoming circuits
On 5/28/23 14:20, Steve Matzura wrote: Who controls how many times an incoming call from an external (DID) provider will ring before Asterisk picks up the call and handles it internally Asterisk and this is defined with your timeout on the dial command, mine is 26 seconds so around 5 rings. 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
[asterisk-users] Question on ring count on incoming circuits
Who controls how many times an incoming call from an external (DID) provider will ring before Asterisk picks up the call and handles it internally--the provider or Asterisk? If it's the DID provider, I'll work on that with them; if it's Asterisk, I didn't find anything anywhere that looks like it has anything to do with incoming ring count unless you set up a ring-no-answer system. For my purposes, that would mean defining a dummy extension that has no hardware attached to it that would fail over to my current call handling code after it rings once. Is this the proper method for handling this? You might wonder why I wouldn't want a call to a system that simply plays a message and then takes an optional voicemail message to pick up immediately. Short answer: Don't ask (groan). It's what the project supporter wants, presumably so that the person calling into the system will know their call went through and to be ready to hear the outgoing message, I don't know, it's a customer request so I feel duty-bound to figure it out and implement it. -- _ -- 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] Question on ARI externalMedia
Please disregard, I figured out what I was doing wrong.
Dan
From: Dan Cropp
Sent: Friday, January 20, 2023 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
password_format = plain
password = Password
I send this curl -v -u MyApp:Password -X POST
"http://localhost:8088/ari/channels/externalMedia?channelId=1234abcd5678=MyApp_host=192.168.33.32%3A1053=slin16;
I can make other ARI commands work, so it must be something specific to my
externalMedia command and the parameters.
The output is the following...
{"id":"1234abcd5678","name":"UnicastRTP/192.168.33.32:1053-0x7fcffc020300","state":"Down","protocol_id":"","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-01-20T10:59:24.569-0600","language":"en","channelvars":{"UNICASTRTP_LOCAL_PORT":"19194","UNICASTRTP_LOCAL_ADDRESS":"192.168.33.31"}}
Dan
--
_
-- 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] Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
password_format = plain
password = Password
I send this curl -v -u MyApp:Password -X POST
"http://localhost:8088/ari/channels/externalMedia?channelId=1234abcd5678=MyApp_host=192.168.33.32%3A1053=slin16;
I can make other ARI commands work, so it must be something specific to my
externalMedia command and the parameters.
The output is the following...
{"id":"1234abcd5678","name":"UnicastRTP/192.168.33.32:1053-0x7fcffc020300","state":"Down","protocol_id":"","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-01-20T10:59:24.569-0600","language":"en","channelvars":{"UNICASTRTP_LOCAL_PORT":"19194","UNICASTRTP_LOCAL_ADDRESS":"192.168.33.31"}}
Dan
--
_
-- 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] Question on resources
On Mon, Sep 5, 2022 at 9:16 AM Mark Murawski wrote: > On 8/4/22 20:32, Jerry Geis wrote: > > I am running Asterisk 13.30.0 > > 40 core CPU (VM) VMware. > > CentOS 7 > > 32 G ram > > 10G vmx network > > > > Should be plenty of room for anything... > > > > Yes asterisk is running 270% CPU... > > Is it not taking advantage of the 40 cores ? > > I am bring around 300 SIP endpoints in a muted audio conference (so > > one way) and this spikes up the CPU to 270%. > > > > Is there something I dont have set right to take advantage to > > the resourses? > > Thanks > > > > Jerry > > > > Hi Jerry, > > If I recall correctly, there was a talk at an AstriCon or a web page > somewhere that I came across at one point (I'm having a hard time > finding it now) that dove in fairly deep into Asterisk performance > related to multiple cores. > > And if I recall correctly, the conclusion was that the drop-off was > around 8-12 cores -- and beyond that the extra cores aren't doing much > other than helping schedule work and you can't really get more > concurrent calls by adding more cores. > > Someone who is a bit more well-versed in large-machine performance with > Asterisk can certainly chime in here, but from what I gather, throwing > 40 cores at a single Asterisk instance is not the magic bullet to > support a massive number of calls. > > > Thanks Mark, Jerry -- _ -- 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] Question on resources
On 8/4/22 20:32, Jerry Geis wrote: I am running Asterisk 13.30.0 40 core CPU (VM) VMware. CentOS 7 32 G ram 10G vmx network Should be plenty of room for anything... Yes asterisk is running 270% CPU... Is it not taking advantage of the 40 cores ? I am bring around 300 SIP endpoints in a muted audio conference (so one way) and this spikes up the CPU to 270%. Is there something I dont have set right to take advantage to the resourses? Thanks Jerry Hi Jerry, If I recall correctly, there was a talk at an AstriCon or a web page somewhere that I came across at one point (I'm having a hard time finding it now) that dove in fairly deep into Asterisk performance related to multiple cores. And if I recall correctly, the conclusion was that the drop-off was around 8-12 cores -- and beyond that the extra cores aren't doing much other than helping schedule work and you can't really get more concurrent calls by adding more cores. Someone who is a bit more well-versed in large-machine performance with Asterisk can certainly chime in here, but from what I gather, throwing 40 cores at a single Asterisk instance is not the magic bullet to support a massive number of calls. -- _ -- 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] Question on Originate with EarlyMedia
On Thu, Sep 1, 2022 at 1:32 PM Dan Cropp wrote: > Using AMI, we send an Originate with EarlyMedia: true setting > > > > If the other end sends a 183, Asterisk > > When the 183 is received, Asterisk indicates the ChannelState: 6 and > ChannelStateDesc: Up values. > > All is fine up to this point. > > > > It may take the caller several seconds before the called party answers. > > When the called party answers (200 OK received), in the debugging I see > Asterisk processing this and debugging show TSX State: Terminated Inv > State: EARLY > > At this point, the call is truly connected. > > > > Is there a configuration setting to indicate whether Asterisk should send > an event indicating when the early media ends and the call is really Up? > There is no option. The Originate code makes the channel appear as answered when the 183 arrives, everything reflects that afterwards. Even if a second answer occurs it gets ignored. The log message you refer to is internal state information to do with the SIP side. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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
[asterisk-users] Question on Originate with EarlyMedia
Using AMI, we send an Originate with EarlyMedia: true setting If the other end sends a 183, Asterisk When the 183 is received, Asterisk indicates the ChannelState: 6 and ChannelStateDesc: Up values. All is fine up to this point. It may take the caller several seconds before the called party answers. When the called party answers (200 OK received), in the debugging I see Asterisk processing this and debugging show TSX State: Terminated Inv State: EARLY At this point, the call is truly connected. Is there a configuration setting to indicate whether Asterisk should send an event indicating when the early media ends and the call is really Up? Dan -- _ -- 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] Question on resources
Hi, Am Donnerstag, dem 04.08.2022 um 20:32 -0400 schrieb Jerry Geis: > I am running Asterisk 13.30.0 > 40 core CPU (VM) VMware. > CentOS 7 > 32 G ram > 10G vmx network > > Should be plenty of room for anything... > > Yes asterisk is running 270% CPU... > Is it not taking advantage of the 40 cores ? > I am bring around 300 SIP endpoints in a muted audio conference (so > one way) and this spikes up the CPU to 270%. What type of conference? Is it meetme or confbridge? AFAIK meetme is working on a single thread... > > Is there something I dont have set right to take advantage to > the resourses? > Thanks > > Jerry HTH, Karsten -- _ -- 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] Question on resources
Doesn’t that mean, effectively that you are using the equivalent of 100% of 2.7 CPUs? --Don From: asterisk-users On Behalf Of Jerry Geis Sent: Thursday, August 4, 2022 7:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question on resources I am running Asterisk 13.30.0 40 core CPU (VM) VMware. CentOS 7 32 G ram 10G vmx network Should be plenty of room for anything... Yes asterisk is running 270% CPU... Is it not taking advantage of the 40 cores ? I am bring around 300 SIP endpoints in a muted audio conference (so one way) and this spikes up the CPU to 270%. Is there something I dont have set right to take advantage to the resourses? Thanks Jerry -- _ -- 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] Question on resources
I am running Asterisk 13.30.0 40 core CPU (VM) VMware. CentOS 7 32 G ram 10G vmx network Should be plenty of room for anything... Yes asterisk is running 270% CPU... Is it not taking advantage of the 40 cores ? I am bring around 300 SIP endpoints in a muted audio conference (so one way) and this spikes up the CPU to 270%. Is there something I dont have set right to take advantage to the resourses? Thanks Jerry -- _ -- 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] Question about the Geo Location support being added
On Wed, Jul 27, 2022 at 11:02 AM Dan Cropp wrote:
> Looking at the Asterisk wiki
>
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
>
Just FYI, I'm in the process of clarifying and adding more info. Should be
done Friday.
>
>
> I see the dial plan support the GeolocProfileCreate and there is support
> for GEOLOC_PROFILE settings to be set on the dial plan.
>
>
>
> We currently use AMI Originate support. We may have dozens/hundreds of
> calls in the system and external to Asterisk, someone executes a behavior
> where we perform the Originate, if the party answers, we ConfBridge the
> necessary calls together. It can be multiple calls and we never know when
> the total calls bridged together will need to be increased. Because of the
> random increase in calls, we can’t use the Dial to bridge the parties
> together.
>
>
>
> The GEO Location information for the original caller can vary
> significantly because they could be WebRTC. We are planning to require the
> setup of the Geo Location for each call to be provided to us (either via
> the incoming call or it may be provided from third party software). Either
> way, we will know what the GEO Location to use for the Originate. Trying
> to wrap my head around the best way to achieve this.
>
A real scenario to test!!! Thanks!
>
>
> Using AMI Originate, is it possible to set the GEOLOC_PROFILE settings via
> the Variable header?
>
I've not tested this but you don't need to do it at all...
>
>
> My thought would be to configure an outgoing Geo Location profile for the
> PJSIP endpoint, but it would have the minimum settings.
>
Actually it would have a template specifying replacement channel variables.
When sending the AMI Originate, provide all the adjustments to the
> GEOLOC_PROFILE settings via the Variable.
>
>
>
> Is this possible or might there be a better way to achieve this?
>
>
>
It's possible but probably not needed. Let's say you're using Civic
Address and a direct originate to the remote party via Dial. In the
originate, you can specify regular, inherited channel variables with the
official Civic Address parameters preceded by '_'. Let's use HNO (house
number) as an example. You'd set _HNO=1633 in the originate and since it
has the '_' prefix it's going to be inherited by the outgoing channel. In
the outgoing channel's profile/location, you'd set 'location_info =
HNO=${_HNO}. Of course there'd be more than just the HNO parameter set but
it's the same technique. The outgoing channel has a very generic location
template populated with values received from the incoming channel.
Now, this isn't going to work if you're originating both calls and adding
them to a bridge yourself but in this case, you have both channels at the
same time so you can just add the incoming channel's location info
directly to the outgoing channel's variables as you originate the outgoing
call. Youdon';t need to create a new GEOLOC_PROFILE for the outgoing
channel.
All of this assumes that I actually understood your situation correctly. :)
How are you getting the caller's info in the first place?
>
>
> Alternatively, I could generate an internal local channel, configure the
> GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then
> have it perform the Dial. If the other end answers or not, treat it
> exactly as we currently do using the Originate.
>
Sounds more complicated than it needs to be.
>
>
>
>
> Dan
>
>
> --
> _
> -- 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
--
_
-- 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] Question about the Geo Location support being added
Looking at the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation I see the dial plan support the GeolocProfileCreate and there is support for GEOLOC_PROFILE settings to be set on the dial plan. We currently use AMI Originate support. We may have dozens/hundreds of calls in the system and external to Asterisk, someone executes a behavior where we perform the Originate, if the party answers, we ConfBridge the necessary calls together. It can be multiple calls and we never know when the total calls bridged together will need to be increased. Because of the random increase in calls, we can't use the Dial to bridge the parties together. The GEO Location information for the original caller can vary significantly because they could be WebRTC. We are planning to require the setup of the Geo Location for each call to be provided to us (either via the incoming call or it may be provided from third party software). Either way, we will know what the GEO Location to use for the Originate. Trying to wrap my head around the best way to achieve this. Using AMI Originate, is it possible to set the GEOLOC_PROFILE settings via the Variable header? My thought would be to configure an outgoing Geo Location profile for the PJSIP endpoint, but it would have the minimum settings. When sending the AMI Originate, provide all the adjustments to the GEOLOC_PROFILE settings via the Variable. Is this possible or might there be a better way to achieve this? Alternatively, I could generate an internal local channel, configure the GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then have it perform the Dial. If the other end answers or not, treat it exactly as we currently do using the Originate. Dan -- _ -- 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] Question on ExternalMedia and the codec
On Tue, Oct 12, 2021 at 2:54 PM Dan Cropp wrote: > We tell asterisk to use the slin format for ExternalMedia. However, the > unicast channel is selecting ulaw formatand the RTP data is indicating it’s > ulaw format. > > > > Anyone know why ulaw format would be on chosen? > What do your ARI requests look like? Are you just requesting "slin" or one of the specific variants? > > > > > [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is > /ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569_host=192.168.32.148:8080 > =slin > > [10/12 16:13:39.396] DEBUG[1665] http.c: match request > [ari/channels/externalMedia] with handler [static] len 6 > > [10/12 16:13:39.396] DEBUG[1665] http.c: match request > [ari/channels/externalMedia] with handler [ari] len 3 > > [10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari] > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for > channels/externalMedia > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari > deviceStates: Didn't match channels > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari > applications: Didn't match channels > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari > channels: Explicit match with channels > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for > externalMedia > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels > create: Didn't match externalMedia > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels > channelId: Matched wildcard. > > [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels > externalMedia: Explicit match with externalMedia > > [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', > our source address is '192.168.33.34'. > > [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for > RTP instance '0x7fef60018320' > > [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP > allocated port 12226 > > [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE > creating session 192.168.33.34:12226 (12226) > > [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE > create > > [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE > add system candidates > > [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE > add candidate: 192.168.33.34:12226, 2130706431 > > [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance > '0x7fef60018320' is setup and ready to go > > [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: : > Formats: (none) > > [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: Channel is being > initialized or destroyed > > [10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name: > channel:1634055219.4, detail: > > [10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4': > 0x7fef6008d170 created > > [10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910 > 'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated > > [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', > our source address is '192.168.33.34'. > > [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: > UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw) > > [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: New topology set > > [10/12 16:13:39.396] DEBUG[1665] res_stasis.c: > a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4 > > [10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1 > interested in a2519b4b-4d90-4d18-906b-717d02f8d569 > > [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open. > status_code:200 > > [10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2 > interested in a2519b4b-4d90-4d18-906b-717d02f8d569 > > > > Have a good day! > > Dan > > This email is intended only for the use of the party to which it is > addressed and may contain information that is privileged, confidential, or > protected by law. If you are not the intended recipient you are hereby > notified that any dissemination, copying or distribution of this email or > its contents is strictly prohibited. If you have received this message in > error, please notify us immediately by replying to the message and deleting > it from your computer. > -- > _ > -- 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] Question on ExternalMedia and the codec
We tell asterisk to use the slin format for ExternalMedia. However, the unicast channel is selecting ulaw formatand the RTP data is indicating it's ulaw format. Anyone know why ulaw format would be on chosen? [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is /ari/channels/externalMedia?app=a2519b4b-4d90-4d18-906b-717d02f8d569_host=192.168.32.148:8080=slin [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [static] len 6 [10/12 16:13:39.396] DEBUG[1665] http.c: match request [ari/channels/externalMedia] with handler [ari] len 3 [10/12 16:13:39.396] DEBUG[1665] http.c: Match made with [ari] [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels/externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari deviceStates: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari applications: Didn't match channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking ari channels: Explicit match with channels [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Finding handler for externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels create: Didn't match externalMedia [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels channelId: Matched wildcard. [10/12 16:13:39.396] DEBUG[1665] res_ari.c: Checking channels externalMedia: Explicit match with externalMedia [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fef60018320' [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) RTP allocated port 12226 [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE creating session 192.168.33.34:12226 (12226) [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE create [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add system candidates [10/12 16:13:39.396] DEBUG[1665] res_rtp_asterisk.c: (0x7fef60018320) ICE add candidate: 192.168.33.34:12226, 2130706431 [10/12 16:13:39.396] DEBUG[1665] rtp_engine.c: RTP instance '0x7fef60018320' is setup and ready to go [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: : Formats: (none) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: Channel is being initialized or destroyed [10/12 16:13:39.396] DEBUG[1665] stasis.c: Creating topic. name: channel:1634055219.4, detail: [10/12 16:13:39.396] DEBUG[1665] stasis.c: Topic 'channel:1634055219.4': 0x7fef6008d170 created [10/12 16:13:39.396] DEBUG[1665] channel.c: Channel 0x7fef6008a910 'UnicastRTP/192.168.32.148:8080-0x7fef60018320' allocated [10/12 16:13:39.396] DEBUG[1665] acl.c: For destination '192.168.32.148', our source address is '192.168.33.34'. [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: UnicastRTP/192.168.32.148:8080-0x7fef60018320: Formats: (ulaw) [10/12 16:13:39.396] DEBUG[1665] channel_internal_api.c: New topology set [10/12 16:13:39.396] DEBUG[1665] res_stasis.c: a2519b4b-4d90-4d18-906b-717d02f8d569: Subscribing to 1634055219.4 [10/12 16:13:39.396] DEBUG[1665] stasis/app.c: Channel '1634055219.4' is 1 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 [10/12 16:13:39.396] DEBUG[1665] http.c: HTTP keeping session open. status_code:200 [10/12 16:13:39.396] DEBUG[1666] stasis/app.c: Channel '1634055219.4' is 2 interested in a2519b4b-4d90-4d18-906b-717d02f8d569 Have a good day! Dan This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. -- _ -- 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] Question on pjsip.conf and aors
Thanks Joshua From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, February 14, 2020 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on pjsip.conf and aors On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp mailto:d...@amtelco.com>> wrote: I have the following configuration… [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a register attempt is received, asterisk outputs… [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found for endpoint '1004' If I change the aor3 to be 1004, everything works. As in [aor3] becomes [1004] and in the endpoint change aors = aor3 to be aors = 1004 Is there a setting I’m missing to allow the endpoint named 1004 to use an auth that doesn’t have the same 1004 name? There isn't a configuration option. AOR is a SIP concept, and in fact when you send a REGISTER you state which AOR you are registering to. Your REGISTER is therefore saying "add me to AOR 1004". Since it's not saying "add me to aor3" it doesn't work. Some devices allow you to specify while others just assume that everything uses your username. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://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] Question on pjsip.conf and aors
On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp wrote: > I have the following configuration… > > > > [aor3] > > type = aor > > max_contacts = 1 > > remove_existing = yes > > > > [auth3] > > type = auth > > username = 1004 > > password = SuperSecretProbation > > > > [1004] > > type = endpoint > > context = IS > > transport = transport1 > > auth = auth3 > > aors = aor3 > > accountcode = 3 > > dtmf_mode = rfc4733 > > device_state_busy_at = 2 > > force_rport = no > > moh_passthrough = yes > > disallow = all > > allow = ulaw > > acl = acl1 > > > > > > When a register attempt is received, asterisk outputs… > > [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not > found for endpoint '1004' > > > > If I change the aor3 to be 1004, everything works. As in [aor3] becomes > [1004] and in the endpoint change aors = aor3 to be aors = 1004 > Is there a setting I’m missing to allow the endpoint named 1004 to use an > auth that doesn’t have the same 1004 name? > There isn't a configuration option. AOR is a SIP concept, and in fact when you send a REGISTER you state which AOR you are registering to. Your REGISTER is therefore saying "add me to AOR 1004". Since it's not saying "add me to aor3" it doesn't work. Some devices allow you to specify while others just assume that everything uses your username. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and 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
[asterisk-users] Question on pjsip.conf and aors
I have the following configuration... [aor3] type = aor max_contacts = 1 remove_existing = yes [auth3] type = auth username = 1004 password = SuperSecretProbation [1004] type = endpoint context = IS transport = transport1 auth = auth3 aors = aor3 accountcode = 3 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no moh_passthrough = yes disallow = all allow = ulaw acl = acl1 When a register attempt is received, asterisk outputs... [02/14 12:53:29.870] WARNING[7883] res_pjsip_registrar.c: AOR '1004' not found for endpoint '1004' If I change the aor3 to be 1004, everything works. As in [aor3] becomes [1004] and in the endpoint change aors = aor3 to be aors = 1004 Is there a setting I'm missing to allow the endpoint named 1004 to use an auth that doesn't have the same 1004 name? Dan -- _ -- 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] Question on WebRTC configuration
Hello, Reading this old thread, isn't there also an error in [1] as It also mentions a tlscafile setting. Cheers [1] https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone Le ven. 7 déc. 2018 à 16:41, Kevin Harwell a écrit : > On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp wrote: > >> In the asterisk wiki instructions for Configuring Asterisk for WebRTC >> clients… >> >> >> >> >> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients >> >> >> >> “To communicate with websocket clients, Asterisk uses its built-in HTTP >> daemon. Configure */etc/asterisk/http.conf* as follows: >> >> >> >> [general] >> >> enabled=yes >> >> bindaddr=0.0.0.0 >> >> bindport=8088 >> >> tlsenable=yes >> >> tlsbindaddr=0.0.0.0:8089 >> >> tlscertfile= >> >> tlsprivatekey= >> >> tlscafile=” >> >> >> >> What is the tlscafile setting? >> >> >> >> When I look at the http.conf samples it doesn’t mention the tlscafile >> setting. >> >> I see there is a tlscafile setting in sip.conf, but I don’t find this >> anywhere else. >> >> >> >> Is the wiki web page mistaken or is this an actual http.conf setting that >> is undocumented? >> > > The page is mistaken. It should not be there. the 'tlscafile' option is > not supported by the Asterisk http server. I've removed it from the wiki. > Thanks for catching that! > > >> >> >> Have a great day! >> > > You too! > > >> Dan >> -- >> > > -- > Kevin Harwell > Digium - A Sangoma Company | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: https://digium.com & https://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 -- _ -- 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] Question on calculating PJSIP md5 authentication with NEC
Thank you Joshua. We are confident the problem is with NEC. One day, Asterisk PJSIP REGISTER response (md5) was being rejected by NEC. Next morning, it's suddenly working with no changes to asterisk. Same exact configuration settings. Suddenly, last Friday NEC starts rejecting the REGISTER again with no changes to asterisk configuration file. NEC has since responded that they have a separate PJSIP setting for REGISTRATION. We are trying to find out more information from them. NEC claim doesn't explain why it worked for several hours and suddenly stopped working. This really feels like they have been modifying settings on their end without informing us. Dan -Original Message- From: asterisk-users On Behalf Of Joshua C. Colp Sent: Monday, July 15, 2019 1:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Question on calculating PJSIP md5 authentication with NEC On Fri, Jul 12, 2019, at 5:10 PM, Dan Cropp wrote: > > Just tracked down the code for the chan_sip MD5 REGISTER and have been > able to verify that chan_sip is calculating the HA1 same as I am > calculating the md5_cred for PJSIP > > 3016:a...@xyz.com:3016 > > > Both chan_sip and PJSIP REGISTER traces show the uri as > sip:10.100.102.82 > > So the HA2 for both as MD5(method:uri) (which in this case is > REGISTER:sip:10.100.102.82). > > > chan_sip response formula with qop detected (sent by NEC) is > > ha1:nonce:noncecount:cnonce:auth:h2 > > Using this formula I am able to match it with my Asterisk chan_sip trace. > > NEC accepts this REGISTER > > > Can anyone point me to the area where Asterisk PJSIP would be doing > the response for the REGISTER 401 reply? > > NEC does not like the way PJSIP calculates the response value for the > REGISTER reply. > > I tried to calculate the response exactly like chan_sip does (using > the values from the trace and the HA1/HA2) but it’s not matching what > the sip trace shows Asterisk sending for the response. > > > Does Asterisk PJSIP support handle this or is it all done by PJSIP? The value is given to PJSIP[1] and it does the rest. You could use the password option and trace how PJSIP is calculating it in that scenario, and then switch to MD5 cred after you determine that and make it match. [1] https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_authenticator_digest.c#L180 -- 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 -- _ -- 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] Question on calculating PJSIP md5 authentication with NEC
On Fri, Jul 12, 2019, at 5:10 PM, Dan Cropp wrote: > > Just tracked down the code for the chan_sip MD5 REGISTER and have been > able to verify that chan_sip is calculating the HA1 same as I am > calculating the md5_cred for PJSIP > > 3016:a...@xyz.com:3016 > > > Both chan_sip and PJSIP REGISTER traces show the uri as sip:10.100.102.82 > > So the HA2 for both as MD5(method:uri) (which in this case is > REGISTER:sip:10.100.102.82). > > > chan_sip response formula with qop detected (sent by NEC) is > > ha1:nonce:noncecount:cnonce:auth:h2 > > Using this formula I am able to match it with my Asterisk chan_sip trace. > > NEC accepts this REGISTER > > > Can anyone point me to the area where Asterisk PJSIP would be doing the > response for the REGISTER 401 reply? > > NEC does not like the way PJSIP calculates the response value for the > REGISTER reply. > > I tried to calculate the response exactly like chan_sip does (using the > values from the trace and the HA1/HA2) but it’s not matching what the > sip trace shows Asterisk sending for the response. > > > Does Asterisk PJSIP support handle this or is it all done by PJSIP? The value is given to PJSIP[1] and it does the rest. You could use the password option and trace how PJSIP is calculating it in that scenario, and then switch to MD5 cred after you determine that and make it match. [1] https://github.com/asterisk/asterisk/blob/master/res/res_pjsip_authenticator_digest.c#L180 -- 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] Question on calculating PJSIP md5 authentication with NEC
Just tracked down the code for the chan_sip MD5 REGISTER and have been able to verify that chan_sip is calculating the HA1 same as I am calculating the md5_cred for PJSIP 3016:a...@xyz.com:3016 Both chan_sip and PJSIP REGISTER traces show the uri as sip:10.100.102.82 So the HA2 for both as MD5(method:uri) (which in this case is REGISTER:sip:10.100.102.82). chan_sip response formula with qop detected (sent by NEC) is ha1:nonce:noncecount:cnonce:auth:h2 Using this formula I am able to match it with my Asterisk chan_sip trace. NEC accepts this REGISTER Can anyone point me to the area where Asterisk PJSIP would be doing the response for the REGISTER 401 reply? NEC does not like the way PJSIP calculates the response value for the REGISTER reply. I tried to calculate the response exactly like chan_sip does (using the values from the trace and the HA1/HA2) but it's not matching what the sip trace shows Asterisk sending for the response. Does Asterisk PJSIP support handle this or is it all done by PJSIP? Dan From: asterisk-users On Behalf Of Dan Cropp Sent: Friday, July 12, 2019 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on calculating PJSIP md5 authentication with NEC I have done additional testing and I haven't been able to figure out why it's failing. Since my original testing we now set the realm on the authentication section to match what we receive from NEC. It's of the format a...@xyz.com<mailto:a...@xyz.com> I have verified the md5_cred several times and it matches the user:realm:password formula 3016:ins...@something0a64.com:3016 where username is 3016 and password is 3016 We suspect it has something to do with the format of the realm that NEC is sending where it may not be working correctly supported by the Asterisk PJSIP code. Is there anyone who has used PJSIP outbound REGISTRATION using MD5 support that can provide some insight? Or even anyone who know chan_sip's REGISTER and how it calculates it's HA1, HA2 for the MD5 authentication? Dan From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Behalf Of Dan Cropp Sent: Wednesday, July 10, 2019 10:48 AM To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com> Subject: [asterisk-users] Question on calculating the md5_sum Using chan_sip, we are able to register with an NEC switch. When I try to REGISTER with PJSIP, the authentication is being rejected. Traces show it's using md5 authentication. The packets looks almost identical. The one area that I suspect is causing the problem is the md5_cred for my pjsip.conf registration. I'm using a Poco MD5 utility to generate the MD5 passing username:realm:password Where username is 3016 Realm is asterisk (default) Password is 3016 which is the same as chan_sip's secret The value I'm setting the md5_cred in auth section to is 63e8aedc77335879c93123055d21211d Would this value match what chan_sip would pass as the md5 credentials? Our sip.conf looks like the following... [general] context = NECTEST bindaddr = 0.0.0.0 bindport = 5060 websocket_enabled = false srvlookup = no allowguest = yes debug = yes sipdebug = yes defaultexpiry = 480 deny = 0.0.0.0/24 permit = 10.100.102.0/24 permit = 192.168.9.0/24 canreinvite = yes callcounter = yes register = 3016:3016@10.100.102.82:5060/3016 [3016] type = friend qualify = no nat = no host = 10.100.102.82:5060 defaultuser = 3016 secret = 3016 incominglimit = 24 accountcode = 33 port = 5060 context = NECTEST dtmfmode = auto disallow = all allow = ulaw defaultexpiry = 480 insecure = invite fromdomain = 10.100.102.82 acl = acl6 Have a great day! Dan -- _ -- 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] Question on calculating PJSIP md5 authentication with NEC
I have done additional testing and I haven't been able to figure out why it's failing. Since my original testing we now set the realm on the authentication section to match what we receive from NEC. It's of the format a...@xyz.com I have verified the md5_cred several times and it matches the user:realm:password formula 3016:ins...@something0a64.com:3016 where username is 3016 and password is 3016 We suspect it has something to do with the format of the realm that NEC is sending where it may not be working correctly supported by the Asterisk PJSIP code. Is there anyone who has used PJSIP outbound REGISTRATION using MD5 support that can provide some insight? Or even anyone who know chan_sip's REGISTER and how it calculates it's HA1, HA2 for the MD5 authentication? Dan From: asterisk-users On Behalf Of Dan Cropp Sent: Wednesday, July 10, 2019 10:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Question on calculating the md5_sum Using chan_sip, we are able to register with an NEC switch. When I try to REGISTER with PJSIP, the authentication is being rejected. Traces show it's using md5 authentication. The packets looks almost identical. The one area that I suspect is causing the problem is the md5_cred for my pjsip.conf registration. I'm using a Poco MD5 utility to generate the MD5 passing username:realm:password Where username is 3016 Realm is asterisk (default) Password is 3016 which is the same as chan_sip's secret The value I'm setting the md5_cred in auth section to is 63e8aedc77335879c93123055d21211d Would this value match what chan_sip would pass as the md5 credentials? Our sip.conf looks like the following... [general] context = NECTEST bindaddr = 0.0.0.0 bindport = 5060 websocket_enabled = false srvlookup = no allowguest = yes debug = yes sipdebug = yes defaultexpiry = 480 deny = 0.0.0.0/24 permit = 10.100.102.0/24 permit = 192.168.9.0/24 canreinvite = yes callcounter = yes register = 3016:3016@10.100.102.82:5060/3016 [3016] type = friend qualify = no nat = no host = 10.100.102.82:5060 defaultuser = 3016 secret = 3016 incominglimit = 24 accountcode = 33 port = 5060 context = NECTEST dtmfmode = auto disallow = all allow = ulaw defaultexpiry = 480 insecure = invite fromdomain = 10.100.102.82 acl = acl6 Have a great day! Dan -- _ -- 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] Question on calculating the md5_sum
Using chan_sip, we are able to register with an NEC switch. When I try to REGISTER with PJSIP, the authentication is being rejected. Traces show it's using md5 authentication. The packets looks almost identical. The one area that I suspect is causing the problem is the md5_cred for my pjsip.conf registration. I'm using a Poco MD5 utility to generate the MD5 passing username:realm:password Where username is 3016 Realm is asterisk (default) Password is 3016 which is the same as chan_sip's secret The value I'm setting the md5_cred in auth section to is 63e8aedc77335879c93123055d21211d Would this value match what chan_sip would pass as the md5 credentials? Our sip.conf looks like the following... [general] context = NECTEST bindaddr = 0.0.0.0 bindport = 5060 websocket_enabled = false srvlookup = no allowguest = yes debug = yes sipdebug = yes defaultexpiry = 480 deny = 0.0.0.0/24 permit = 10.100.102.0/24 permit = 192.168.9.0/24 canreinvite = yes callcounter = yes register = 3016:3016@10.100.102.82:5060/3016 [3016] type = friend qualify = no nat = no host = 10.100.102.82:5060 defaultuser = 3016 secret = 3016 incominglimit = 24 accountcode = 33 port = 5060 context = NECTEST dtmfmode = auto disallow = all allow = ulaw defaultexpiry = 480 insecure = invite fromdomain = 10.100.102.82 acl = acl6 Have a great day! Dan -- _ -- 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] Question about packet counts in voipmonitor
Mike, Are you using port mirroring or is VoipMonitor running on the same box? If the latter I would run tcpdump and compare what VM says it has to what you see in your wireshark dump. If you are sniffing via port mirroring your switch maybe dropping packets (we had that when we tried to mirror 700 mbit of traffic on a Juniper EX4200). On Fri, Dec 21, 2018 at 8:13 PM Mike Diehl wrote: > Hi all, > > > > I'm not sure this is the place to ask, but here goes... > > > > I'm using voipmonitor to gather call statistics such as packet counts, > average jitter, etc. Eventually, I want to use those stats to detect and > alert on poor call quality. > > > > However, I'm finding that the packet counts for each leg of a given call > can vary quite a bit. > > > > For example, I have a call that was connected for 84 seconds. At 50 > frames/sec, I expect to see about 4200 frames. However, on one side I see > 4187 (which is good) and on the other side, I only see 2577 frames sent. > > > > Am I doing something wrong? Or is this approach simply doomed? > > > > Any thoughts would be welcome. > > > > -- > > Mike Diehl > > > > > -- > _ > -- 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 -- _ -- 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] Question about packet counts in voipmonitor
Hi all, I'm not sure this is the place to ask, but here goes... I'm using voipmonitor to gather call statistics such as packet counts, average jitter, etc. Eventually, I want to use those stats to detect and alert on poor call quality. However, I'm finding that the packet counts for each leg of a given call can vary quite a bit. For example, I have a call that was connected for 84 seconds. At 50 frames/sec, I expect to see about 4200 frames. However, on one side I see 4187 (which is good) and on the other side, I only see 2577 frames sent. Am I doing something wrong? Or is this approach simply doomed? Any thoughts would be welcome. -- Mike Diehl -- _ -- 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] Question on WebRTC configuration
On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp wrote: > In the asterisk wiki instructions for Configuring Asterisk for WebRTC > clients… > > > > > https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients > > > > “To communicate with websocket clients, Asterisk uses its built-in HTTP > daemon. Configure */etc/asterisk/http.conf* as follows: > > > > [general] > > enabled=yes > > bindaddr=0.0.0.0 > > bindport=8088 > > tlsenable=yes > > tlsbindaddr=0.0.0.0:8089 > > tlscertfile= > > tlsprivatekey= > > tlscafile=” > > > > What is the tlscafile setting? > > > > When I look at the http.conf samples it doesn’t mention the tlscafile > setting. > > I see there is a tlscafile setting in sip.conf, but I don’t find this > anywhere else. > > > > Is the wiki web page mistaken or is this an actual http.conf setting that > is undocumented? > The page is mistaken. It should not be there. the 'tlscafile' option is not supported by the Asterisk http server. I've removed it from the wiki. Thanks for catching that! > > > Have a great day! > You too! > Dan > -- > -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://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
[asterisk-users] Question on WebRTC configuration
In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients... https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients "To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows: [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile= tlsprivatekey= tlscafile=" What is the tlscafile setting? When I look at the http.conf samples it doesn't mention the tlscafile setting. I see there is a tlscafile setting in sip.conf, but I don't find this anywhere else. Is the wiki web page mistaken or is this an actual http.conf setting that is undocumented? Have a great day! Dan -- _ -- 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] Question on PJSIP's endpoint section in wiki
On Fri, Apr 27, 2018, at 11:13 AM, Olivier wrote: > Hello, > > I don't know if this list is the best place to ask such question but here > it is, anyway. > > In page [1], I can read in PJSIP's endpoint section configuration reference: > identify_by username,location Way(s) for Endpoint to be > identified > > Then clicking over identify_by text, you can read: > identify_by ... supported options are username, ... and auth_username > > How do yopu read it ? > I would expect the first line to written as: > dentify_by username,auth_username Way(s) for Endpoint to be > identified The wiki documentation hasn't been regenerated lately (it's in queue to be fixed). "username,auth_username" would be correct. There's also others[1] depending on version. [1] https://github.com/asterisk/asterisk/blob/13/configs/samples/pjsip.conf.sample#L633 -- Joshua Colp Digium, Inc. | 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
[asterisk-users] Question on PJSIP's endpoint section in wiki
Hello, I don't know if this list is the best place to ask such question but here it is, anyway. In page [1], I can read in PJSIP's endpoint section configuration reference: identify_by username,location Way(s) for Endpoint to be identified Then clicking over identify_by text, you can read: identify_by ... supported options are username, ... and auth_username How do yopu read it ? I would expect the first line to written as: dentify_by username,auth_username Way(s) for Endpoint to be identified Thoughts ? Best regards [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_identify_by -- _ -- 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] Question on permit/deny
On 1 July 2015 at 04:03, Jerry Geis ge...@pagestation.com wrote: I see in my log file this: Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' ( 5.189.144.120:5076) to extension '011972592675431' rejected because extension not found in context 'default'. which is great its rejected - however in my sip.conf file I have deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255 So I'm expecting to deny everything and only allow the two addresses I have listed of which the 5.189.144.120 is not one of? What is wrong with my permit/deny ? Thanks, Jerry -- _ Check your sip.conf to see if allowguest is explicitly set to no. ;context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ; If your Asterisk is connected to the Internet ; and you have allowguest=yes ; you want to check which services you offer everyone ; out there, by enabling them in the default context (see below). Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on permit/deny
I see in my log file this: Jun 30 21:44:26] NOTICE[42192][C-02f3] chan_sip.c: Call from '' ( 5.189.144.120:5076) to extension '011972592675431' rejected because extension not found in context 'default'. which is great its rejected - however in my sip.conf file I have deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255 So I'm expecting to deny everything and only allow the two addresses I have listed of which the 5.189.144.120 is not one of? What is wrong with my permit/deny ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about hangup - Asterisk v11.15.0
Hello,
on previous versions of asterisk, extension h and H make us know who
ended a call (caller or callee). In the last * versions, seems that only
h extension is used, as stated here
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
In the last versions, how do we know which end terminate a call (SIP,
ISDN, Analog, ...) in h extension ? Will the
${HASH(SIP_CAUSE,${CDR(dstchannel)})} give the information ?
We also face a strange behavior: we are ringing few phones (~10) and
sometimes, once the call get answered, we see that 2~3 seconds after
this, music on hold is started on the channel! And 20 seconds after, the
call is terminated without that any party hanged up :-(
It's a Elastix 2.5 installation, we thought that problem could came from
Elastix so we set our own dialplan for incoming calls:
same =
n,Set(__phonesToRing=SIP/118SIP/119SIP/122SIP/123SIP/124SIP/125SIP/126SIP/127SIP/128SIP/129SIP/130SIP/132)
same = n(startRing),Answer()
same = n,Dial(${phonesToRing},,it) ;no voicemail
or forward = ring indefenitely
same = n,Hangup
Incoming call give for instance in logs:
[2015-03-23 11:07:20] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/126-43d8 is ringing
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/118-43d3 connected line has changed. Saving it until answer for
SIP/bero_trunk-43d2
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: --
SIP/118-43d3 answered SIP/bero_trunk-43d2
[2015-03-23 11:07:25] VERBOSE[1342][C-0e85] res_musiconhold.c:
-- Started music on hold, class 'default', on SIP/bero_trunk-43d2
[2015-03-23 11:07:27] VERBOSE[1342][C-0e85] res_musiconhold.c:
-- Stopped music on hold on SIP/bero_trunk-43d2
[2015-03-23 11:07:41] VERBOSE[1342][C-0e85] pbx.c: -- Executing
[h@from-trunk:1] Macro(SIP/bero_trunk-43d2, hangupcall,) in new
stack
Thanks for any hint
--
Daniel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function sip_write inside chan_sip.c.
In our previous version (11.2.1) we did not see those messages being printed
(same verbosity level). We compared both versions of the functions and see no
difference at all in the 'default' switch case that handles that. We
think/assume that that function is being called in
different places on each version (11.2-1 vs 13-1).
We also think it has to do with the asterisk receiving rtp packets with comfort
noise which is not supported by asterisk.
We would like to know what can we do about it to behave more like the version
11?
We are not sure but could it be that version 11 handles it better ?. I am
attaching the functions on both versions for your review.
Thank you
/*! \brief Send frame to media channel (rtp) */
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast_channel_tech_pvt(ast);
int res = 0;
switch (frame-frametype) {
case AST_FRAME_VOICE:
if
(!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast),
frame-subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, Asked to transmit frame type %s,
while native formats is %s read/write = %s/%s\n,
ast_getformatname(frame-subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast_channel_nativeformats(ast)),
ast_getformatname(ast_channel_readformat(ast)),
ast_getformatname(ast_channel_writeformat(ast)));
return 0;
}
if (p) {
sip_pvt_lock(p);
if (p-t38.state == T38_ENABLED) {
/* drop frame, can't sent VOICE frames while in
T.38 mode */
sip_pvt_unlock(p);
break;
} else if (p-rtp) {
/* If channel is not up, activate early media
session */
if ((ast_channel_state(ast) != AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
!ast_test_flag(p-flags[0], SIP_OUTGOING))
{
ast_rtp_instance_update_source(p-rtp);
if (!global_prematuremediafilter) {
p-invitestate =
INV_EARLY_MEDIA;
transmit_provisional_response(p, 183 Session Progress, p-initreq, TRUE);
ast_set_flag(p-flags[0],
SIP_PROGRESS_SENT);
}
}
p-lastrtptx = time(NULL);
res = ast_rtp_instance_write(p-rtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_VIDEO:
if (p) {
sip_pvt_lock(p);
if (p-vrtp) {
/* Activate video early media */
if ((ast_channel_state(ast) != AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
!ast_test_flag(p-flags[0], SIP_OUTGOING))
{
p-invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, 183
Session Progress, p-initreq, TRUE);
ast_set_flag(p-flags[0],
SIP_PROGRESS_SENT);
}
p-lastrtptx = time(NULL);
res = ast_rtp_instance_write(p-vrtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_TEXT:
if (p) {
sip_pvt_lock(p);
if (p-red) {
ast_rtp_red_buffer(p-trtp, frame);
} else {
if (p-trtp) {
/* Activate text early media */
if ((ast_channel_state(ast) !=
AST_STATE_UP)
!ast_test_flag(p-flags[0],
SIP_PROGRESS_SENT)
Re: [asterisk-users] Question about Warning message
thank you, we are using the same configuration files in 13, same setup, just different asterisk version. we just dont see the msgs in the console/logs, it is the same exact voice traffic on both asterisk versions is that something that you set on/off? if that is the case how can it be done? what is the alternative? what are their differences/characteristics? how to choose one over among others? thank you again From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: Question about Warning message Date: Mon, 23 Feb 2015 12:27:05 -0500 Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our logs and console: WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type frames with SIP write) We found that line in function sip_write inside chan_sip.c. In our previous version (11.2.1) we did not see those messages being printed (same verbosity level). We compared both versions of the functions and see no difference at all in the 'default' switch case that handles that. We think/assume that that function is being called in different places on each version (11.2-1 vs 13-1). We also think it has to do with the asterisk receiving rtp packets with comfort noise which is not supported by asterisk. We would like to know what can we do about it to behave more like the version 11? We are not sure but could it be that version 11 handles it better ?. I am attaching the functions on both versions for your review. Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. This is basically what we do, we use puppet to help distribute files to remote servers while still using app_queue. Shared network drive also works. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
Oops, quite right, how typoful of me! Thanks for the excellent points, I'll look into gluster and puppet and see may way onwards from there. cheers, Olli 2015-02-06 12:32 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk: On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers
On 06/02/15 07:54, Olli Heiskanen wrote: My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? I assume you mean NFS. Yes you can do that although using NFS you will then have a single point of failure and in the standard NFS client configuration if you try to access a file which is on NFS but it is unavailable then the file access will hang. So you might be better off having the files copied onto each of the asterisks servers local file storage or use a redundant file system such as gluster. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regarding custom announcements used by several Asterisk servers
Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on multicast source
I have a machine with three IP addresses. NIC eth0 NIC eth1 and a virtual address on ETH1 All my devices work normally communicating to the virtual address on eth1. My question is just for mulitcast. The end device has an option for allowed source so I put in the virtual address from my server. No multicast audio received... I then disabled the allowed source and tried again. I received multicast audio. My question is how do I set on Asterisk 11.15.0 the source address for multicasting? In my sip.conf I do have the bind parameter set to my virtual address. Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SIP warning
Hi, upto asterisk 1.8 you used to get this error if there were more than 1 m= line in an invite... Asterisk was just telling you it was declining the second. I belive from 10.0 onwards asterisk now just replies back with port 0 to the stream it isn't interested in... You can ignore it - if its bothering you upgrade to asterisk 11 which is very solid now. On 6 September 2014 10:28, CDR vene...@gmail.com wrote: I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream: audio 30660 RTP/AVP 4 101 13 What does it mean and does it show a problem like one-way audio? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SIP warning
I get tons of these messages chan_sip.c:10088 process_sdp: Declining non-primary audio stream: audio 30660 RTP/AVP 4 101 13 What does it mean and does it show a problem like one-way audio? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about SIP Dial
It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss)
Regards
On Fri, Aug 15, 2014 at 6:20 AM, CDR vene...@gmail.com wrote:
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
but it does not work. I just cannot embed the IP address in the peer's
definition, but I need to use some other configuration features that
are unique to each peer.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about SIP Dial
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
but it does not work. I just cannot embed the IP address in the peer's
definition, but I need to use some other configuration features that
are unique to each peer.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about PJSIP
I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command pjsip reload was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PJSIP
On Mon, Jul 21, 2014 at 7:00 PM, CDR vene...@gmail.com wrote: I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command pjsip reload was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong? We routinely run multiple Asterisk instances on a single machine using the PJSIP stack. A log showing messages why the res_pjsip_* modules couldn't be loaded on a particular instance of Asterisk would be helpful. -- Matthew Jordan Digium, Inc. | Engineering Manager 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk 12
Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a little lost. I see that Asterisk 12 has a nice REST API - very nice - something I can use. However, and this is gonna sound dumb - but all the CLI commands are different now. What did I miss? Can anyone, please, anyone point me to a good, simple to understand tutorial on the new CLI? I am so, so freaking lost! I'm not looking for hand-holding, I just want to understand. Something that will show me how to: - create users - configure SIP trunks - configure basic dialplan I'm lost - anyone point me to a resource that is easy to follow? Once I get the jist, I think I'll be fine. I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? Overwhelming - sigh. Thank much - any help would be appreciated - next time you are in Manchester NH - I'll make you my fave Tequila Sour drink! G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk 12
Maybe it's just me if I'm not mistaken the three things you listed are usually configured using the config files not on CLI. Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Wystead Sent: Wednesday, January 22, 2014 3:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about Asterisk 12 Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a little lost. I see that Asterisk 12 has a nice REST API - very nice - something I can use. However, and this is gonna sound dumb - but all the CLI commands are different now. What did I miss? Can anyone, please, anyone point me to a good, simple to understand tutorial on the new CLI? I am so, so freaking lost! I'm not looking for hand-holding, I just want to understand. Something that will show me how to: * create users * configure SIP trunks * configure basic dialplan I'm lost - anyone point me to a resource that is easy to follow? Once I get the jist, I think I'll be fine. I looked on http://www.voip-info.org http://www.voip-info.org/ - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? Overwhelming - sigh. Thank much - any help would be appreciated - next time you are in Manchester NH - I'll make you my fave Tequila Sour drink! G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Asterisk 12
I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? https://wiki.asterisk.org/wiki/display/AST/Home To my knowledge the voip-info domain is mostly outdated information these days. The official asterisk wiki is where you want to look for current information. Sorry I can't help you with anything else, I've not had time to play with 12 Yet. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about Management Interface
CDR wrote: I am trying to identify the module (*.so) that contains the Asterisk Management Interface, so as to set noload=XXX.so in modules.conf. Any idea? There is no module, it's provided as core functionality. Disabling it can be done in manager.conf -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Management Interface
I am trying to identify the module (*.so) that contains the Asterisk Management Interface, so as to set noload=XXX.so in modules.conf. Any idea? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
Jonas Kellens wrote: Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 for RTCP. If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? Yes. I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Yes. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about how Asterisk works with RTP ports
Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Thanks. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
On 10/29/2013 05:14 PM, Joshua Colp wrote: Jonas Kellens wrote: Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 for RTCP. If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? Yes. I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Yes. So if I understand correct, you don't need to look at the amount of concurrent calls to calculate the RTP range in rtp.conf, you need to look at the amount of INVITES that are being send at one moment ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
Jonas Kellens wrote: So if I understand correct, you don't need to look at the amount of concurrent calls to calculate the RTP range in rtp.conf, you need to look at the amount of INVITES that are being send at one moment ? The number of concurrent channels in existence which are using RTP. While a channel may not be answered, it's still in existence. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question Asterisk Manager
Hi A small question on Asterisk Manager. I use Perl Script for start a call: my $response = $astman-sendcommand( Action = 'Originate', Channel = 'SIP/ASTERISK/$Extension', Exten = '200', Context = 'MyContext', Priority = '1', Async = '1' ); That's start the call, but only the position of the corresponding sounds departing. As soon as he clinched, that the second ringing phone. Is there a way for two phone ring at the same time? Thanks Olivier -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on AEL2 string comparisons
On Thu, Jul 4, 2013 at 12:24 AM, James B. Byrne byrn...@harte-lyne.cawrote:
I have this code in a dial plan:
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41799]?notfromlocal)
exten = _417XX,n,GotoIf($[${CALLERID(num)}
SIP/41700]?notfromlocal)
The value of ${CALLERID(num)} appears to be SIP/41712-0181
-- Executing [41720@from-internal:5] GotoIf(SIP/41712-0181,
0?notfromlocal) in new stack
-- Executing [41720@from-internal:6] GotoIf(SIP/41712-0181,
1?notfromlocal) in new stack
-- Goto (from-internal,41720,8
This value is evidently comparing to be less than SIP/41799 as
expected but also is considered less than SIP/41700 as well, which
is not expected (by me). What am I doing wrong here?
What I am attempting to accomplish is to detect calls originally made
from internal extension numbers in the range 41700..41799 inclusive.
What is the correct method to accomplish this?
James B. Byrne
${CALLERID(num)} should give you only number and not technology i.e. 41712.
Give this a shot,
exten = _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten = _417XX,n,GotoIf($[$[${CALLERID(num)} 41799] |
$[${CALLERID(num)} 41700]]?notfromlocal:)
--Satish Barot
Ahmedabad, India
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on AEL2 string comparisons
On Thu, July 4, 2013 02:14, Satish Barot wrote:
${CALLERID(num)} should give you only number and not technology i.e.
41712.
Give this a shot,
exten = _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten = _417XX,n,GotoIf($[$[${CALLERID(num)} 41799] |
$[${CALLERID(num)} 41700]]?notfromlocal:)
--Satish Barot
Ahmedabad, India
That works. Thank you.
--
*** E-Mail is NOT a SECURE channel ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte Lyne Limited http://www.harte-lyne.ca
9 Brockley Drive vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada L8E 3C3
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about media before connect
I need to block any audio before there is a connect, in SIP. How do I tell the DIAL application to behave like that? Yours Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question
Is it me or Google just blocked Asterisk's chan_motif? I get violation of terms of service audio message whenever I send a call. Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question
CDR wrote: Is it me or Google just blocked Asterisk's chan_motif? I get violation of terms of service audio message whenever I send a call. Works fine here. Their automated security system probably determined your usage behavior was not consistent with normal usage and terminated your access. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about CDR
thanks asghar for your help and support and thanks ishfaq 2013/5/9 Asghar Mohammad asghar...@gmail.com hi, asterisk insert cdr when call is hangup and last dial statment, i dont understatnd why you are using 2 dial statment on same extenstion? if you you want dial to both extensions you can use 506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want to do failover the check Dial status and gotoif dialstatus = NO ANSWER or what ever you need. On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about CDR
hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about CDR
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203| 506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 Hi You need to look into Channel Event Logging https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932 Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about CDR
thanks i verify but i don't understanding if can someone give me an example best regards 2013/5/9 Ishfaq Malik i...@pack-net.co.uk On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203| 506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 Hi You need to look into Channel Event Logging https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932 Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about CDR
hi, asterisk insert cdr when call is hangup and last dial statment, i dont understatnd why you are using 2 dial statment on same extenstion? if you you want dial to both extensions you can use 506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want to do failover the check Dial status and gotoif dialstatus = NO ANSWER or what ever you need. On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
On Mon, Mar 25, 2013 at 03:15:24PM +, Salaheddine Elharit wrote: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers Asterisk 1.4 is at build time set to use either DAHDI or Zaptel (but not both). (try: 'strings /usr/sbin/asterisk | grep /dev'). So you'll have to at least rebuild Asterisk vs. DAHDI. Asterisk of older versions does not support DAHDI at all. You should also note that even the branch 1.4.x is no longer actively supported, and this would be a good time to upgrade. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command /etc/asterisk/zapata.conf is a configuration ifle of Asterisk's chan_zap.so alone. So changes to it would generally require no more than restart of Asterisk. The simpler of them would be applied with a simple reload (or 'reload chan_zap.so' as you mention). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
ok thanks for your help and support i really appreciated 2013/3/26 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Mar 25, 2013 at 10:44:47AM +, Salaheddine Elharit wrote: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command /etc/asterisk/zapata.conf is a configuration ifle of Asterisk's chan_zap.so alone. So changes to it would generally require no more than restart of Asterisk. The simpler of them would be applied with a simple reload (or 'reload chan_zap.so' as you mention). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about zapata.conf
hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . “service zaptel restart” or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de mailto:yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
ok thank you so much for your help and support 2013/3/25 Yves A. yves...@gmx.de hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
The Dial events are created by app_dial. So long as you are using app_dial to create your outbound channel, you should have that event. Channel technology shouldn't matter. I am using the same AMI method to start both calls. Action: Originate Channel: DAHDI/18/XX or Action: Originate Channel: SIP/machine/XX Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2) did my AMI call Action: Originate Async: yes Channel: SIP/testsystem/XXX (calls from my machine over SIP trunk to another 11.0.2 box that has a PRI card to make a call out to my cell) and did not get a break. Why is a SIP call not logging the Dial event as a DAHDI call does??? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on SIP trunk and AMI to place call
When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d And get the dialed number from that. Actually you could issue the AMI command core show channels verbose. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Sent: Thursday, January 24, 2013 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call? some of them are bound to have the variable callerid set yes I have looked at all of them, CallerID is not set to the number I am calling. Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
Not the greatest solution, but since you are most likely using a script for the AMI process, you could do an Asterisk --rx core show channels verbose|grep SIP/testmachine-000d And get the dialed number from that. Actually you could issue the AMI command core show channels verbose. there is no core show channels verbose on Asterisk 11. There is on asterisk 1.4, core show channels on asterisk 11 has been changed. jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
On 01/24/2013 10:46 AM, Jerry Geis wrote: When I am monitoring the AMI I see the following event for a call I just made over a SIP trunk. Event: Newchannel Privilege: call,all Channel: SIP/testmachine-000d ChannelState: 0 ChannelStateDesc: Down CallerIDNum: CallerIDName: AccountCode: Exten: Context: testmachine Uniqueid: 1359035395.20 In this event or any event following I do not see the phone number that I dialled. How do I correlate the SIP/testmachine-000d to the number I just dialed (purpose is to hangup the call later if I need to interrupt it) Now if I am using a machine with actual hardware cards, the phone number is included as part of the Channel so I can look that up. but for a SIP trunk the phone number dialled does not come over the AMI. How do I match up the call I just started (using AMI over SIP trunk) to the number I called? You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. For example: exten = 500,1,Dial(SIP/digium02) Results in: Event: Dial Privilege: call,all SubEvent: Begin Channel: SIP/10.x.x.x-0002 Destination: SIP/digium02-0003 CallerIDNum: 657-5309 CallerIDName: digium01 ConnectedLineNum: unknown ConnectedLineName: unknown UniqueID: Asterisk-01-1359052866.2 DestUniqueID: Asterisk-01-1359052866.3 Dialstring: digium02 -- Matthew Jordan Digium, Inc. | Engineering Manager 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
This might have changed but IIRC /etc/asterisk/manager.conf controls what events you have access to. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, January 24, 2013 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on SIP trunk and AMI to place call
On 01/24/2013 01:13 PM, Jerry Geis wrote: You probably want the Dial event. It is raised both at the beginning of the Dial, as well as when the Dial completes. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial Note that the Channel: field will contain the name initiating the Dial, the Destination: field will contain the channel name being dialled, and the Dialstring: field will contain the non-technology specific portion of the thing being dialled. I get that even on the system with the PRI card and using DAHDI however I am not getting that event on the system with the SIP trunk . Is there something to enable to get that??? Both systems are running Asterisk 11.0.2. The Dial events are created by app_dial. So long as you are using app_dial to create your outbound channel, you should have that event. Channel technology shouldn't matter. -- Matthew Jordan Digium, Inc. | Engineering Manager 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about directmedia or canreinvite in sip.conf
Hello, I have a question about directmedia or canreinvite, I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from sip show settings that my directmedia configuration is applied. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question on Confbridge menu item dialplan_exec
I like the example of using that to add somebody to the conference, but what I don't see is how the dialplan can know what conference the menu item was called from. I was hoping that some variable might have been set, but don't see it in the sources. Is the idea to do that outside of the call to Confbridge? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
