[asterisk-users] Log to file in Asterisk: append with newline
Hi all, (I am testing on Asterisk 11.7.0~dfsg-1ubuntu1) I am using the following format to append to a logfile, according to the documentation http://www.voip-info.org/wiki/view/Asterisk+func+FILE: same = n,Set(FILE(/tmp/mylog.txt,,,a)=my-log-message) But this does not append a newline. So I am trying: same = n,Set(FILE(/tmp/mylog.txt,,,al)=my-log-message) But this does not append (instead, it just overwrites - surprisingly, since the documentation says that this should append!). How can I append *and* make sure that a newline is added after the log message? Thanks, Daniel Gonzalez -- _ -- 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] Passing literals with commas to subroutine
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do not know if the variable
has a comma or not.)
Thanks,
Daniel
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[asterisk-users] chanspy (whispering) and Mixmonitor
Dear all, i'm using Chanspy to dynamically play a sound file on a specific channel. It works the caller and the callee can hear the file playing during their conversation. However, i'm also using Mixmonitor to record the call. The thing is, in the resulting wav i can of course hear the conversation, but not the sound which was whispering. Anyone knows how to let the whisper being recorded please ? regards, José -- _ -- 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] Want web page to listen to meetme (WebRTC?)
Quick and drity: 1) Meetme has to be configured to record the media stream. 2) You have to install a streaming server. Maybe ffmpeg could do the job: https://trac.ffmpeg.org/wiki/StreamingGuide 3) Then your website should be able to get the stream from the streaming server. You should be able to test this scenario withing some hours. Am 08.12.2014 16:11, schrieb Steve Edwards: I have a web page to do the usual meetme admin stuff -- mute, kick, etc. Now, the client is asking if they can listen to the meetme -- click and audio comes out the computer speakers. How can this be implemented? Is this a use case for WebRTC? -- _ -- 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] Planned maintenance for community services on Tuesday, 9th of December 2014
Tonight several community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 9:00 PM CST[1] and should last no longer than three hours, ending around 12:00 AM CST. The affected services are: * JIRA issue tracker (issues.asterisk.org) * Proxy for various community services, including but not limited to: ** wiki.asterisk.org ** reviewboard.asterisk.org ** code.asterisk.org * Downloads sites, including, but not limited to: ** downloads.asterisk.org ** downloads.digium.com ** packages.asterisk.org ** asterisknow.asterisk.org) Thank you for your support! [1]: http://tinyurl.com/lfydmuw (see converted times) -- Digium's Asterisk Development Team 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] Playing audio to bridged channels using ControlPlayBack
One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 – 125 active users. The ultimate goal is several hundred concurrent users and I can see that happening with this app. Much of the functionality comes from a PERL script that runs as a background daemon. I think that’s a plus, because I can easily add additional features by modifying the script. Since I knew I would be making a lot of modifications to the daemon script, I ended up refactoring the whole thing. Write me if you want a copy. In my opinion, if you are going to use app_konference, there is no need to use ControlPlayBack since the script will do that for you. Once of the drawbacks is that the discussion group is pretty much dead, so if you run into a problem, there's really nowhere to go. Overall, I think this is a good application. I hope this helps. Regards; John V. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Monday, December 08, 2014 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Saturday, December 06, 2014 8:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Playing audio to bridged channels I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is interested. Your help is appreciated. Thanks Murthy -- _ -- 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] Issue between Asterisk Queue and GSM gateway when trying to use call waiting feature
El Lunes, 8 de diciembre, 2014 12:51:42, Matthew Jordan mjor...@digium.com escribió: On Fri, Dec 5, 2014 at 3:23 PM, Rodrigo Montiel guevara2...@yahoo.com.ar wrote: Hi masters, I’m not an expert on this my friends, but I’m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario: Caller — GSM Gateway with SIM card A — Asterisk queue — extension 1000 GSM gateway with call waiting activated on SIM A Queue with “skip busy agent” disabled and ringall strategy. SIP extension 1000 with call waiting activated, and member of Asterisk queue. a) One Caller calls to SIM card A of GSM Gateway, call is forwarded to Asterisk queue where SIP extension 1000 answers. b) New Caller calls the same SIM card A of GSM gateway (it has call waiting activated on the sim card), call is forwarded to Asterisk queue to the same extension 1000 and a pop-up appears with the second call. c) extension 1000 accepts it so put on hold first call, then try to pickup the new one. The thing is that the SIP re-invite with sendonly attribute can be seen from extension 1000 to Asterisk queue, but this SIP invite is not being forwarded to GSM gateway. So the GSM gateway keeps waiting for it and because it never appears the 1st call is dropped. Maybe you have had this issue in the past. I know that Im not an expert, but I have been researching a lot and trying to vary configurations without clues. The question is: Is it expected for the Asterisk queue to redirect this on-hold message (SIP re-invite with sendonly media attribute) to the GSM gateway so it can manage it call waiting feature on the same SIM card? If we repeat the same scenario without queue intervention (i.e. call goes directly to the extension) the SIP re-invite floods normally between Asterisk and GSM gateway, so GSM gateway can decide what to do with the call. I have no specific queue configuration, seems that queues.conf does not have any parameter to allow this behaviour of re-sending re-invite/on-hold messages. Vendor from GSM gateway side is pointing that “Asterisk js not resending on-hold message”. Asterisk is a back to back user agent. As such, it does not forward or proxy any SIP messages. The re-INVITE sent from the SIP device represented by extension 1000 in your scenario is handled by Asterisk, and causes the channel on the other side of the bridge with that SIP channel to be put on Hold. There is no mechanism in Asterisk today to pass through a re-INVITE to initiate a remote hold. -- 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 Thank you master!! -- _ -- 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] Asterisk 12 - Security Fix Only Notice
On Monday, December 08, 2014 02:21:16 PM Matthew Jordan wrote: Hey everyone! This is a friendly reminder that Asterisk 12 will be entering security fix only mode soon. As a Standard release of Asterisk, Asterisk 12 received one year of maintenance fixes, and will receive one year of security fixes. Asterisk 12 was first released on 2013-12-20 - the one year anniversary of which is just around the corner! After 2014-12-20, additional releases of Asterisk 12 will no longer be made. The final bug fix release of Asterisk 12 will therefore be 12.8.0. Users of Asterisk 12 are encouraged to move to the next major version, Asterisk 13, as soon as possible. Asterisk 13 is a Long Term Support (LTS) and has maintenance support through 2018-10-24, with its full End of Life occurring on 2019-10-24. For more information on Asterisk versions and their supported lifetimes, please see the following wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Thank you for your continued support of Asterisk! Is there any time frame for when FFA will be available for 13? -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701 -- _ -- 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] Passing literals with commas to subroutine
On Tuesday 09 Dec 2014, Daniel Gonzalez wrote:
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do not know if the variable
has a comma or not.)
What happens if you use speech marks around the variable, like so: ${data} ?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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[asterisk-users] higher cpu usage 1.8 - 11
hi, i upgraded few asterisk systems from last asterisk 1.8 to 11 and i see in graph that cpu usage is ~50% higher any ideas? configuration, modules, .. is the same -- --- Marek Cervenka === -- _ -- 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] Playing audio to bridged channels using ControlPlayBack
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support Sent: Tuesday, December 09, 2014 7:01 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 – 125 active users. The ultimate goal is several hundred concurrent users and I can see that happening with this app. Much of the functionality comes from a PERL script that runs as a background daemon. I think that’s a plus, because I can easily add additional features by modifying the script. Since I knew I would be making a lot of modifications to the daemon script, I ended up refactoring the whole thing. Write me if you want a copy. In my opinion, if you are going to use app_konference, there is no need to use ControlPlayBack since the script will do that for you. Once of the drawbacks is that the discussion group is pretty much dead, so if you run into a problem, there's really nowhere to go. Overall, I think this is a good application. I hope this helps. Regards; John V. Hi John Going through App Konference sources, the following seemed interesting 1) the sounds are packaged in the module--specifically asterikast/audio/join.wav and asterikast/audio/leave.wav (I would like to choose the sound file to play dynamically) 2)the sound controls are mute/unmute, volume up/down (I would like forwarding, pause, etc. that are available with ControlPlayBack) 3)the sound files are selectively played to a channel (unlike multicasting) Is it the listener.pl the daemon script you refer to? A cursory glance at it shows that it is possible to call any command supported by the konference/cli.c. Could you tell me what customizations you have made? I would like, of course, to see your version of the script. Please either post in this thread or send me the script, whichever is easier. My email address is in the header. Thanks and looking forward to hearing from you. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Monday, December 08, 2014 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Playing audio to bridged channels using ControlPlayBack There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota Sent: Saturday, December 06, 2014 8:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Playing audio to bridged channels I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is interested. Your help is appreciated. Thanks Murthy -- _ -- 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] Bridge configuration in Asterisk 13
Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue. Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says MOH will no longer work in softmix, you should use simple. This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.? Regards, Patrick. -- _ -- 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] About voip gateway
I want to create a voip service, I do not know much about it, but the first thing I want to know if more than one client can make a call at the same time through internet to the PSTN, and what gateway should I use for this. 2014-12-08 13:07 GMT-08:00 Steve Edwards asterisk@sedwards.com: On Mon, 8 Dec 2014, Leonel Florin wrote: Hay friends, I want to know how many simultaneous call can i do throughout a voip gateway from the internet call to the normal telephony network, because i want to see what implementation do i have to do multiple call from internet to differents telephones. Please reply with a few more details of what you are planning on doing. For example: I want my computer to originate 100 simultaneous calls to PSTN subscribers who have 'opted-in' to receive a 60 second political announcement.' If all you want to do is route calls, OpenSIPS may be a better tool. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Bridge configuration in Asterisk 13
On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont p.beaum...@hatsoffsoftware.co.uk wrote: Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1t=92007 I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue. Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says MOH will no longer work in softmix, you should use simple. This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly. The bridging technology bridge_softmix is only used by app_confbridge in Asterisk v11. Nothing else in v11 uses the bridging framework. Unless you were using app_confbridge, you were not using bridge_softmix in v11. The various bridging technology modules in v12 and later are for different scenarios. The bridging framework is smart enough to pick the best bridging technology available for the situation. If the situation changes during a call, the bridging framework can change the bridge technology to support the new situation. * bridge_simple is for normal two party communication. * bridge_native_rtp is a special case of two party bridge were both parties use RTP for media exchange. The native technology allows for direct media. * bridge_softmix is for multi-party bridges where you can have 1 to n users communicating in a conference. As you found out, bridge_softmix can be used as a fallback if bridge_simple is not available because it allows two party communication. * bridge_holding is a parking bridge technology to hold calls for later connection. Parties in a holding bridge cannot communicate with each other. * bridge_builtin_features and bridge_builtin_interval_feature provide functionality used by features.conf. These two modules are actually not bridging technologies but support code for features.conf functionality. You usually need to install all of the bridging technologies. Richard -- _ -- 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] Bridge configuration in Asterisk 13 [Spam score:8%]
Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent bridge_simple but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Richard Mudgett rmudg...@digium.com Sent: 09 December 2014 20:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridge configuration in Asterisk 13 [Spam score:8%] On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont p.beaum...@hatsoffsoftware.co.ukmailto:p.beaum...@hatsoffsoftware.co.uk wrote: Hi Everyone. I was referred here by malcolmd of the Asterisk forums. What follows is a copy of this question: http://forums.asterisk.org/viewtopic.php?f=1t=92007? I've recently upgraded from Asterisk 11 to Asterisk 13. Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13. The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple. What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue. Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says MOH will no longer work in softmix, you should use simple. This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.? The bridging technology bridge_softmix is only used by app_confbridge in Asterisk v11. Nothing else in v11 uses the bridging framework. Unless you were using app_confbridge, you were not using bridge_softmix in v11. The various bridging technology modules in v12 and later are for different scenarios. The bridging framework is smart enough to pick the best bridging technology available for the situation. If the situation changes during a call, the bridging framework can change the bridge technology to support the new situation. * bridge_simple is for normal two party communication. * bridge_native_rtp is a special case of two party bridge were both parties use RTP for media exchange. The native technology allows for direct media. * bridge_softmix is for multi-party bridges where you can have 1 to n users communicating in a conference. As you found out, bridge_softmix can be used as a fallback if bridge_simple is not available because it allows two party communication. * bridge_holding is a parking bridge technology to hold calls for later connection. Parties in a holding bridge cannot communicate with each other. * bridge_builtin_features and bridge_builtin_interval_feature provide functionality used by features.conf. These two modules are actually not bridging technologies but support code for features.conf functionality. You usually need to install all of the bridging technologies. Richard -- _ -- 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] About voip gateway
I want to create a voip service, I do not know much about it, but the first thing I want to know if more than one client can make a call at the same time through internet to the PSTN, and what gateway should I use for this. I think the first recommendation any of us will have is to research all you can as there are a lot of mistakes to be made in the telephony world and some of them can be expensive and/or dangerous. The kinds of questions you are asking are not bad to ask, but they do place you squarely at a beginner level. It is hard to answer your questions without having further information. What are you trying to accomplish with this system? Do you need to carry more than one call? What types of phone service are available where this will be installed? For example, a single POTS line will allow you one call in or out of the PSTN. This is not a limitation of Asterisk, this is a limitation on how POTS lines work. A PRI style connection (E1 or T1 depending on location) will allow many more (over 20 calls at once). A SIP trunk is only limited by the number of lines your trunking provider allows and the bandwidth of your internet connection. The gateway you would want to use will depend entirely on what type of connection to the PSTN you are using. A lot of manufacturers make hardware compatible with Asterisk for physical connections to the PSTN and a SIP trunk just requires an internet connection of sufficiently high bandwidth, low latency and a reasonably stable path to the SIP provider. Without knowing more about what you are aiming to do, it is hard for anyone to give you any specific help. You were earlier asked for a specific example of what you wish to accomplish. Please provide that and you will get more people responding. -- _ -- 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] Bridge configuration in Asterisk 13 [Spam score:8%]
On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont p.beaum...@hatsoffsoftware.co.uk wrote: Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent bridge_simple but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a normal two party call then MOH would no longer function? That is correct. bridge_softmix is optimized for multi-party conferencing where passing control frames such as hold/unhold to other parties in the bridge is not a good idea. For example, if three parties are in a bridge and if party A pressed its hold button then that should not necessarily prevent parties B and C from talking to each other. Using bridge_softmix for a normal two party call is a last resort. It works reasonably well as a normal two party bridge technology but it is computationally expensive and not intended for that purpose. Richard -- _ -- 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] From external IP am not able hear the audio on the SIP extensions
Hi, In my office have setuped the Elastix machine and i have a static IP (external IP given by ISP), now the issue is that whenerve call from outside sip extensions which is register to the sip server , am not able hear audio from both side. both callee and caller cant hear audio. please help me on this -- Regards Upendra -- _ -- 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
