Re: [asterisk-users] saved.gsm -> Voicemail greeting ??
If you do VoiceMail(bEXTEN) then all you get is "Please leave your message after the tone" so you can do the greeting function in the dialplan. If you do VoiceMail(bEXTEN) then you get " is on the phone. Please leave..." (uEXTEN) gets you " is unavailable. Please leave..." On 9/15/06, Nick Ellson <[EMAIL PROTECTED]> wrote: Trying that now... umm, anyone know what condition makes use of just the "name" in voicemail, is that part of the directory or something? -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Fri, 15 Sep 2006, Bill Gibbs wrote: > I assume it will use the files .gsm too? > > Bill > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs > Sent: Friday, September 15, 2006 10:46 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] saved.gsm -> Voicemail greeting ?? > > Example for mailbox 100 under context default > > /var/spool/asterisk/voicemail/default/100 > > -rwx-w 1 asterisk asterisk 442604 Aug 22 16:44 busy.wav > -rwx-w 1 asterisk asterisk 44976 Aug 22 16:44 busy.WAV > -rwx-w 1 asterisk asterisk 25964 Aug 8 02:17 greet.wav > -rwx-w 1 asterisk asterisk 2660 Aug 8 02:17 greet.WAV > drwx-w 2 asterisk asterisk 4096 Sep 13 13:45 INBOX > drwx-w 2 asterisk asterisk 4096 Sep 13 13:37 tmp > -rwx-w 1 asterisk asterisk 168044 Aug 9 17:00 unavail.wav > -rwx-w 1 asterisk asterisk 17090 Aug 9 17:00 unavail.WAV > drwx-w 2 asterisk asterisk 4096 Sep 1 11:31 Work > > Those are the files (wav format) that it expects for the voicemail > greetings/name announcement. Greet.wav is the name. > > Bill > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Nick > Ellson > Sent: Friday, September 15, 2006 10:26 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] saved.gsm -> Voicemail greeting ?? > > > Hi John, > > Yes, I followed an example that put all my family sound files in > /var/lib/asterisk/sounds/local, which is also where this file is. Now I > am > trying to figure out how to get the unavailable|name|Busy .gsm's I made > loaded into a mailbox without playing my sounds back into a phone ;) > > Nick > > > -- > Nick Ellson > CCDA, CCNP, CCSP, CCAI, > MCSE 2000, Security+, Network+ > Network Hobbyist, VFR Private Pilot. > > > On Fri, 15 Sep 2006, John covici wrote: > >> Check in /var/spool/asterisk/voicemail/default/ for >> a particular extension, don't know how you want to differentiate after >> hours, etc. Also, you can put files in >> /var/lib/asterisk/sounds/custom and do with them what you want. >> >> on Friday 09/15/2006 Nick Ellson([EMAIL PROTECTED]) wrote >>> >>> I seem to have stumped myself on this one. I had my son rattle off > some >>> really great sound bytes for his own extension (busy, after hours, > etc) >>> and that was easy to set up with the dial plan. Now I have his > actual VM >>> greeting in a .gsm and no idea how to get it into his VM Greeting, > I am >>> guessing that these are not stored where the other sounds are, maybe > in >>> the database? I looked through the .pdf book, not many helpful hits > on >>> google. Help? :) >>> >>> Nick >>> >>> -- >>> Nick Ellson >>> CCDA, CCNP, CCSP, CCAI, >>> MCSE 2000, Security+, Network+ >>> Network Hobbyist, VFR Private Pilot. >>> >>> ___ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> Your life is like a penny. You're going to lose it. The question is: >> How do >> you spend it? >> >> John Covici >> [EMAIL PROTECTED] >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailin
Re: [asterisk-users] How to send DTMF down a channel
I followed my intuition and sent a DTMF tone via the D() option in the dial command to the destination and that causes the voice to be transmitted after the call is answered. I also realised that the ringing tone to the destination does not come up until I execute a SendDTMF command before dialling the destination. From that it appears that a command to issue a tone either before dialling out or after the call is picked up is necessary for the exchange of sound to take place, and I don't know whether that is a bug, or a feature of the Asterisk design. In verbose mode the call is interspersed with the output from the complicated AGI script and that will make things difficult, I have the problem (hopefully) solved now, but I will try to make the time to send you the output of the script after I remove the AGI code from it. In dial plan terms the idea is to generate a call file that contains both the caller id of the caller and the destination, and when the caller answers the call is routed to an AGI that extracts the dialled number and dials it as though they were entered via DTMF. And it appears that because no sound stream is generated through entry of DTMF, no sound is transferred after the call is answered. Perhaps trying something like that in a dial plan you create yourself will reproduce the problem. Thanks Frank On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote: could you post the output of the asterisk console in verbose mode? In logger.conf [logfiles] console => notice,warning,error,verbose,debug Regards On 9/15/06, Frank Church <[EMAIL PROTECTED]> wrote: > The program in question is an adaptation an AGI calling card program. > It is adapted for callback by setting by channelling the callback call > into the context used for the normal inbound leg. > > When used that way entering the PIN and destination number via DTMF > works normally. > > I tried to implement sms/web callback simply by passing the > destination number and the callerid in call file's parameters, so that > the script can go straight to dialling the destination number without > prompting the caller for it. > > This method results in dead air when the call is answered at the > destination and it seems that the absence of DTMF tones for generating > the call is the cause. > > Logically it should work, why the call should result in dead air > because the destination number was not obtained through DTMF is what I > can't understand. > > It appears that the absence of DTMF stops the RTP from being > established to the destination number or prevents the calls from > being joined. When I dial back to the calling number I get the busy > voice prompt okay. > > Injecting the DTMF tones on the call channel is a really a kludge to > simulate actual key presses in the hope that the dead air problem will > go away and it . > > On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote: > > Frank. PlayDTMF and SendDTMF is the same as pressing keys at the > > phone. Im not understanding well, can you please explain a practical > > scenario of how do you expect it to work, and how actually works? :) > > Thanks > > > > Regards > > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote: > > > How do you specify the channel the SendDTMF is sent on, as it has to > > > be variable? I am able to use PlayDTMF to send tones using the > > > manager interface - the tones come up okay but it doesn't work. > > > > > > My problem comes from a working callback script that I am trying to > > > adapt to web/SMS callback. > > > > > > The script is unchanged only that in the web/sms callback the > > > destination number is passed in the call file variables to allow it be > > > dialled after the callback is answered rather than being obtained via > > > DTMF. > > > > > > The destination call is made alright, but the called party only gets > > > dead air after the call is answered. It seems that the fact of > > > pressing the DTMF keys activates the sound channel in way that the > > > direct call back does not, which is why I am going through all these > > > hoops. I just hoped that sending some DTMF tones programmatically > > > would simulate the actual key presses. > > > > > > Do you have any ideas of what the problem might be? > > > > > > > > > > > > On 9/14/06, Moises Silva <[EMAIL PROTECTED]> wrote: > > > > http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF > > > > > > > > Regards > > > > > > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote: > > > > > How can DTMF be sent down a channel? > > > > > > > > > > I am thinking of method where say a channel id can be grabbed from > > > > > Asterisk Manager events and a DTMF signal sent down that channel, > > > > > through AGI, Asterisk Manager Interface or whatever? > > > > > > > > > > Is it possible to have a command in extensions.conf which can take > > > > > both the channel and the dtmf numbers as parameters and send the DTMF > > > > > signals? > > > > > > > > > > Frank Church > > > > >
[asterisk-users] read variable from shell script
Hi list, is it possible to call a shell script from * which returns a number or a string which can be read to an asterisk variable? Something like 'Set(VAR(System(/opt/scripts/something.script)))? Does anyone have an idea? Regards, Christophorus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: > I have been searching, but I have not found the answer.. How might I add > the amr codec to my asterisk server? > I believe I found the amr source from > http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip > I compiled it but did not end up with any .so files like I thought I > would need to put it into asterisk. > Any pointers on how to get an amr codec into asterisk would be most > helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] "Ping" a phone
Currently, I send a popup to an agent asking them to accept the call or not. However, they are complaining that they do not *see* the popup (it's onscreen for 7 seconds) and I was wanting (have been asked to provide) a facility where I can "ping" an agents phone before sending the popup. This would make them aware that there is a call about to come in. The best I could come up with so far is a Dial(SIP/Exten,1,g). This seems to work, but I was wondering if there was another more elegant mechanism. We currently use Cisco 7960 phones. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy
Matt, I am sure this is a RTFM and I am pretty sure you are using meetme rooms. Just not too sure how you do the magic. 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups = 665 lines. My client's call volume has shot from 5,000 to about 10,000 calls a day. Due to recent product offerings/advertising, I expect to be eating up 6 T1 (peak) by the end of October. They will eventually have every channel in use during peaks, whether that is in November or December, I am not sure. I just know it can't break at that point due the the sheer expense of revenue lost for downtime. Thanks, Steve Matt Florell wrote: How many lines and agents are you looking at? What kind of call volume? Average expected hold time? VICIDIAL could be an option for you since it does not use Asterisk Queues and can already easily scale across many servers. MATT--- On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote: I have been tossing around some ideas about scaling a call center with load balancing and redundancy and would like the comunities input, thoughts, criticism and anything anyone wants to toss in. The most evident thing is to start with beefy servers and only run procs that are required. All of the TDM boxes run stripped down versions of Linux and Asterisk, they just take the call from the PRIs and convert them to SIP, everything stays ulaw end to end. *Shared queues across multiple servers would be ideal*. I don't think it is possible in asterisk, as is. Maybe DUNDI could be useful but I am not up to speed on it enough to really know. I was toying with a concept of a DB server tracking the number of calls to queue(s), number of agents logged into the queue(s). Some agents will be logged into multiple queues and providing the logic to a series of Asterisk servers. Calls could be made to the db to determine which queue/server to route the call to. In this situation, duplicate queues would exist on several servers, so balancing would work somewhat if the DB made the selection on which box to route the call to and which box an agent should log into. FastAGI and the manager interface will provide the routing and DB updates. Another thought was to have one central server with all of the queues and agents, then somehow the central server would cause a "recording/CDR server" to send re-invites to the two SIP endpoints so that the call/RTP stream is moved to another asterisk server which would record the call and keep the CDR info. Again, this would be done with a DB to decide which asterisk (recording/CDR) box has the lightest load. It would take the burden of maintaining the call from the "Queue" server. I/O is the first bottleneck in scaling when you record each and every call. Would it be difficult to have asterisk send two SIP endpoints re-invites and then bridge the call? Then it is just a matter of the "Queue" server checking the DB which recording/CDR server the call should go to and send it a message to re-invite and bridge the endpoints. A transfer to a meetme is another possiblility but I want the "Queue" server out of the stream. Has anybody else thought through the best way to scale something like this. I have a DS3 and will be using all of the channels in the semi-near future. I need to come up with a workable plan before then. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] "Ping" a phone
How large is the popup? I cant imagine when my outlook 2003 email popup appears (5 seconds I guess) that I don't see it, though I think the movement of it appearing on my display then receeding away has a lot to do with it. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith > Sent: Saturday, 16 September 2006 7:44 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] "Ping" a phone > > Currently, I send a popup to an agent asking them to accept the call or > not. However, they are complaining that they do not *see* the popup > (it's onscreen for 7 seconds) and I was wanting (have been asked to > provide) a facility where I can "ping" an agents phone before sending > the popup. > > This would make them aware that there is a call about to come in. > > The best I could come up with so far is a Dial(SIP/Exten,1,g). This > seems to work, but I was wondering if there was another more elegant > mechanism. We currently use Cisco 7960 phones. > > Julian. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Ping" a phone
Um, it's a 1024x768 bright yellow box ;) I know, unbelievable. Julian. Dean Collins wrote: How large is the popup? I cant imagine when my outlook 2003 email popup appears (5 seconds I guess) that I don't see it, though I think the movement of it appearing on my display then receeding away has a lot to do with it. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Saturday, 16 September 2006 7:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] "Ping" a phone Currently, I send a popup to an agent asking them to accept the call or not. However, they are complaining that they do not *see* the popup (it's onscreen for 7 seconds) and I was wanting (have been asked to provide) a facility where I can "ping" an agents phone before sending the popup. This would make them aware that there is a call about to come in. The best I could come up with so far is a Dial(SIP/Exten,1,g). This seems to work, but I was wondering if there was another more elegant mechanism. We currently use Cisco 7960 phones. Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "Ping" a phone
Julian Lyndon-Smith wrote: Currently, I send a popup to an agent asking them to accept the call or not. However, they are complaining that they do not *see* the popup (it's onscreen for 7 seconds) and I was wanting (have been asked to provide) a facility where I can "ping" an agents phone before sending the popup. This would make them aware that there is a call about to come in. The best I could come up with so far is a Dial(SIP/Exten,1,g). This seems to work, but I was wondering if there was another more elegant mechanism. We currently use Cisco 7960 phones. Julian. You allow your agents to decline calls?!? I think your solution is the simplest way of doing it. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SHSU asterisk installation?
hi all Linked from /. today, http://www.networkworld.com/news/2006/091206- von-sam-houston.html talks about the Sam Houston State University (SHSU) migrating a rather large amount of users to asterisk. The article describes the installation in rather vague terms, so I was wondering if someone know how they plan to do this, in detail. thanks roy --- "Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people" - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SHSU asterisk installation?
Roy Sigurd Karlsbakk wrote: hi all Linked from /. today, http://www.networkworld.com/news/2006/091206-von-sam-houston.html talks about the Sam Houston State University (SHSU) migrating a rather large amount of users to asterisk. The article describes the installation in rather vague terms, so I was wondering if someone know how they plan to do this, in detail. thanks roy --- "Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people" - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] Yes, there have been several threads about this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy
Hello, At this point in time VICIDIAL is more focused on outbound features, but inbound and blended capcbilities have been part of VICIDIAL for about two years. The most I have done inbound-only with it is 3 T1s with 60 agents. But for outbound and inbound agents together we have had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk servers, 2 web servers and 1 MySQL server) handling over 200,000 calls a day(mostly outbound of course). The inbound portion of VICIDIAL does not have customized hold music or periodic announcements yet, but we plan on adding those features in a future version as we begin to focus more on inbound in the project. We do use meetme rooms in VICIDIAL. This allows for easy third-party calls and multiple monitoring/manager intrusion into an agent session. The load balancing works by having the Database keep track of agents and calls for all of the servers and then use IAX to move the calls from where they originated to whoever the next available agent is, no matter what server they are on. So the calls can come from anywhere and the agents can be logged in anywhere. If you receive inbound callerID on these calls you will also be able to have caller information appear on the Agent's web interface. And if they are a customer that is already in the system it will bring up their existing record. As for reliability, we have not had a total system failure in the last 2 years(aside from long power outages and hurricane interruptions). MySQL can handle a tremendous volume and it is the only total-system-single-point-of-failure in VICIDIAL, ours never crashes. The web servers can be load balanced(no need for session-awareness) and you can use any Apache/PHP webserver that may be on your system to serve the AJAX-based agent web interface. As for Asterisk, we have had servers crash periodically(a couple crashes a month across 8 servers), but that is to be expected when you push tens of thousands of calls through each one per day. Will you be recording all calls in this setup? MATT--- On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote: Matt, I am sure this is a RTFM and I am pretty sure you are using meetme rooms. Just not too sure how you do the magic. 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups = 665 lines. My client's call volume has shot from 5,000 to about 10,000 calls a day. Due to recent product offerings/advertising, I expect to be eating up 6 T1 (peak) by the end of October. They will eventually have every channel in use during peaks, whether that is in November or December, I am not sure. I just know it can't break at that point due the the sheer expense of revenue lost for downtime. Thanks, Steve Matt Florell wrote: > How many lines and agents are you looking at? > > What kind of call volume? > > Average expected hold time? > > VICIDIAL could be an option for you since it does not use Asterisk > Queues and can already easily scale across many servers. > > MATT--- > > > On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote: >> I have been tossing around some ideas about scaling a call center with >> load balancing and redundancy and would like the comunities input, >> thoughts, criticism and anything anyone wants to toss in. >> >> The most evident thing is to start with beefy servers and only run procs >> that are required. All of the TDM boxes run stripped down versions of >> Linux and Asterisk, they just take the call from the PRIs and convert >> them to SIP, everything stays ulaw end to end. >> >> *Shared queues across multiple servers would be ideal*. I don't think >> it is possible in asterisk, as is. Maybe DUNDI could be useful but I am >> not up to speed on it enough to really know. >> >> I was toying with a concept of a DB server tracking the number of calls >> to queue(s), number of agents logged into the queue(s). Some agents >> will be logged into multiple queues and providing the logic to a series >> of Asterisk servers. Calls could be made to the db to determine which >> queue/server to route the call to. In this situation, duplicate queues >> would exist on several servers, so balancing would work somewhat if the >> DB made the selection on which box to route the call to and which box an >> agent should log into. FastAGI and the manager interface will provide >> the routing and DB updates. >> >> Another thought was to have one central server with all of the queues >> and agents, then somehow the central server would cause a "recording/CDR >> server" to send re-invites to the two SIP endpoints so that the call/RTP >> stream is moved to another asterisk server which would record the call >> and keep the CDR info. Again, this would be done with a DB to decide >> which asterisk (recording/CDR) box has the lightest load. It would take >> the burden of maintaining the call from the "Queue" server. I/O is the >> first bottleneck in scaling when you record each and every call. >> >> Would it be difficult to hav
Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy
Right now we are all inbound and every call is recorded. Matt Florell wrote: Hello, At this point in time VICIDIAL is more focused on outbound features, but inbound and blended capcbilities have been part of VICIDIAL for about two years. The most I have done inbound-only with it is 3 T1s with 60 agents. But for outbound and inbound agents together we have had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk servers, 2 web servers and 1 MySQL server) handling over 200,000 calls a day(mostly outbound of course). The inbound portion of VICIDIAL does not have customized hold music or periodic announcements yet, but we plan on adding those features in a future version as we begin to focus more on inbound in the project. We do use meetme rooms in VICIDIAL. This allows for easy third-party calls and multiple monitoring/manager intrusion into an agent session. The load balancing works by having the Database keep track of agents and calls for all of the servers and then use IAX to move the calls from where they originated to whoever the next available agent is, no matter what server they are on. So the calls can come from anywhere and the agents can be logged in anywhere. If you receive inbound callerID on these calls you will also be able to have caller information appear on the Agent's web interface. And if they are a customer that is already in the system it will bring up their existing record. As for reliability, we have not had a total system failure in the last 2 years(aside from long power outages and hurricane interruptions). MySQL can handle a tremendous volume and it is the only total-system-single-point-of-failure in VICIDIAL, ours never crashes. The web servers can be load balanced(no need for session-awareness) and you can use any Apache/PHP webserver that may be on your system to serve the AJAX-based agent web interface. As for Asterisk, we have had servers crash periodically(a couple crashes a month across 8 servers), but that is to be expected when you push tens of thousands of calls through each one per day. Will you be recording all calls in this setup? MATT--- On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote: Matt, I am sure this is a RTFM and I am pretty sure you are using meetme rooms. Just not too sure how you do the magic. 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups = 665 lines. My client's call volume has shot from 5,000 to about 10,000 calls a day. Due to recent product offerings/advertising, I expect to be eating up 6 T1 (peak) by the end of October. They will eventually have every channel in use during peaks, whether that is in November or December, I am not sure. I just know it can't break at that point due the the sheer expense of revenue lost for downtime. Thanks, Steve Matt Florell wrote: > How many lines and agents are you looking at? > > What kind of call volume? > > Average expected hold time? > > VICIDIAL could be an option for you since it does not use Asterisk > Queues and can already easily scale across many servers. > > MATT--- > > > On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote: >> I have been tossing around some ideas about scaling a call center with >> load balancing and redundancy and would like the comunities input, >> thoughts, criticism and anything anyone wants to toss in. >> >> The most evident thing is to start with beefy servers and only run procs >> that are required. All of the TDM boxes run stripped down versions of >> Linux and Asterisk, they just take the call from the PRIs and convert >> them to SIP, everything stays ulaw end to end. >> >> *Shared queues across multiple servers would be ideal*. I don't think >> it is possible in asterisk, as is. Maybe DUNDI could be useful but I am >> not up to speed on it enough to really know. >> >> I was toying with a concept of a DB server tracking the number of calls >> to queue(s), number of agents logged into the queue(s). Some agents >> will be logged into multiple queues and providing the logic to a series >> of Asterisk servers. Calls could be made to the db to determine which >> queue/server to route the call to. In this situation, duplicate queues >> would exist on several servers, so balancing would work somewhat if the >> DB made the selection on which box to route the call to and which box an >> agent should log into. FastAGI and the manager interface will provide >> the routing and DB updates. >> >> Another thought was to have one central server with all of the queues >> and agents, then somehow the central server would cause a "recording/CDR >> server" to send re-invites to the two SIP endpoints so that the call/RTP >> stream is moved to another asterisk server which would record the call >> and keep the CDR info. Again, this would be done with a DB to decide >> which asterisk (recording/CDR) box has the lightest load. It would take >> the burden of maintaining the call from the "Queue" server. I/O is the
Re: [asterisk-users] SHSU asterisk installation?
Linked from /. today, http://www.networkworld.com/news/2006/091206- von-sam-houston.html talks about the Sam Houston State University (SHSU) migrating a rather large amount of users to asterisk. The article describes the installation in rather vague terms, so I was wondering if someone know how they plan to do this, in detail. Yes, there have been several threads about this. Obviously I _have_ tried to google about this, so if you could point me to one of those threads, I'd be grateful roy --- "Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people" - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SHSU asterisk installation?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Roy Sigurd Karlsbakk wrote: >>> Linked from /. today, >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html >>> talks about the Sam Houston State University (SHSU) migrating a >>> rather large amount of users to asterisk. The article describes the >>> installation in rather vague terms, so I was wondering if someone >>> know how they plan to do this, in detail. >> >> Yes, there have been several threads about this. > > Obviously I _have_ tried to google about this, so if you could point me > to one of those threads, I'd be grateful - From 3 days ago: http://www.sineapps.com/news.php?rssid=1509 Thread here: [asterisk-users] University switches to Asterisk - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4 KojTXtiqQLK7YNtXp+4Qh+I= =T9d+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SHSU asterisk installation?
Maybe I should just hold a conference call about all this stuff. On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Roy Sigurd Karlsbakk wrote: > >>> Linked from /. today, > >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html > >>> talks about the Sam Houston State University (SHSU) migrating a > >>> rather large amount of users to asterisk. The article describes the > >>> installation in rather vague terms, so I was wondering if someone > >>> know how they plan to do this, in detail. > >> > >> Yes, there have been several threads about this. > > > > Obviously I _have_ tried to google about this, so if you could point me > > to one of those threads, I'd be grateful > > - From 3 days ago: > > http://www.sineapps.com/news.php?rssid=1509 > > Thread here: > > [asterisk-users] University switches to Asterisk > > - -- > Cheers, > > Matt Riddell > ___ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://wap.sineapps.com (Daily Asterisk News for your cellphone) > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.2 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4 > KojTXtiqQLK7YNtXp+4Qh+I= > =T9d+ > -END PGP SIGNATURE- > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] USA Regulatons
Guys, this may be a off question but where do I start to find all the regulations to comply with to set up a residential voip phone service in the US? The tech stuff I am fine with but the legal is where I need some pointers. Also I am not looking for a flame war just pointers on the legal issues. Thanx M -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy
To maintain high recording quality with no audio skips we have found that you should not go over 50 conversations being recorded on a single server. What have you found is your limit while maintaining very good audio quality? MATT--- On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote: Right now we are all inbound and every call is recorded. Matt Florell wrote: > Hello, > > At this point in time VICIDIAL is more focused on outbound features, > but inbound and blended capcbilities have been part of VICIDIAL for > about two years. The most I have done inbound-only with it is 3 T1s > with 60 agents. But for outbound and inbound agents together we have > had upto 120 agents on one setup with 20 T1s(spread across 8 Asterisk > servers, 2 web servers and 1 MySQL server) handling over 200,000 calls > a day(mostly outbound of course). > > The inbound portion of VICIDIAL does not have customized hold music or > periodic announcements yet, but we plan on adding those features in a > future version as we begin to focus more on inbound in the project. > > We do use meetme rooms in VICIDIAL. This allows for easy third-party > calls and multiple monitoring/manager intrusion into an agent session. > > The load balancing works by having the Database keep track of agents > and calls for all of the servers and then use IAX to move the calls > from where they originated to whoever the next available agent is, no > matter what server they are on. So the calls can come from anywhere > and the agents can be logged in anywhere. > > If you receive inbound callerID on these calls you will also be able > to have caller information appear on the Agent's web interface. And if > they are a customer that is already in the system it will bring up > their existing record. > > As for reliability, we have not had a total system failure in the last > 2 years(aside from long power outages and hurricane interruptions). > MySQL can handle a tremendous volume and it is the only > total-system-single-point-of-failure in VICIDIAL, ours never crashes. > The web servers can be load balanced(no need for session-awareness) > and you can use any Apache/PHP webserver that may be on your system to > serve the AJAX-based agent web interface. As for Asterisk, we have had > servers crash periodically(a couple crashes a month across 8 servers), > but that is to be expected when you push tens of thousands of calls > through each one per day. > > Will you be recording all calls in this setup? > > MATT--- > > On 9/16/06, Steve Totaro <[EMAIL PROTECTED]> wrote: >> Matt, >> >> I am sure this is a RTFM and I am pretty sure you are using meetme >> rooms. Just not too sure how you do the magic. >> >> 28 T1s with NFAS so 95 channels per trunk group, seven trunk groups = >> 665 lines. My client's call volume has shot from 5,000 to about 10,000 >> calls a day. Due to recent product offerings/advertising, I expect to >> be eating up 6 T1 (peak) by the end of October. They will eventually >> have every channel in use during peaks, whether that is in November or >> December, I am not sure. I just know it can't break at that point due >> the the sheer expense of revenue lost for downtime. >> >> Thanks, >> Steve >> >> >> Matt Florell wrote: >> > How many lines and agents are you looking at? >> > >> > What kind of call volume? >> > >> > Average expected hold time? >> > >> > VICIDIAL could be an option for you since it does not use Asterisk >> > Queues and can already easily scale across many servers. >> > >> > MATT--- >> > >> > >> > On 9/15/06, Steve Totaro <[EMAIL PROTECTED]> wrote: >> >> I have been tossing around some ideas about scaling a call center >> with >> >> load balancing and redundancy and would like the comunities input, >> >> thoughts, criticism and anything anyone wants to toss in. >> >> >> >> The most evident thing is to start with beefy servers and only run >> procs >> >> that are required. All of the TDM boxes run stripped down >> versions of >> >> Linux and Asterisk, they just take the call from the PRIs and convert >> >> them to SIP, everything stays ulaw end to end. >> >> >> >> *Shared queues across multiple servers would be ideal*. I don't >> think >> >> it is possible in asterisk, as is. Maybe DUNDI could be useful >> but I am >> >> not up to speed on it enough to really know. >> >> >> >> I was toying with a concept of a DB server tracking the number of >> calls >> >> to queue(s), number of agents logged into the queue(s). Some agents >> >> will be logged into multiple queues and providing the logic to a >> series >> >> of Asterisk servers. Calls could be made to the db to determine >> which >> >> queue/server to route the call to. In this situation, duplicate >> queues >> >> would exist on several servers, so balancing would work somewhat >> if the >> >> DB made the selection on which box to route the call to and which >> box an >> >> agent should log into. FastAGI and the manager interface will >> provide >> >> the routing and DB up
Re[2]: [asterisk-users] SHSU asterisk installation?
That would be superb! -Original Message- From: Aaron Daniel <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Sent: Sat, 16 Sep 2006 11:32:00 -0500 Delivered: Sat, 16 Sep 2006 13:21:48 Subject:[asterisk-users] SHSU asterisk installation? Maybe I should just hold a conference call about all this stuff. On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Roy Sigurd Karlsbakk wrote: > >>> Linked from /. today, > >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html > >>> talks about the Sam Houston State University (SHSU) migrating a > >>> rather large amount of users to asterisk. The article describes the > >>> installation in rather vague terms, so I was wondering if someone > >>> know how they plan to do this, in detail. > >> > >> Yes, there have been several threads about this. > > > > Obviously I _have_ tried to google about this, so if you could point me > > to one of those threads, I'd be grateful > > - From 3 days ago: > > http://www.sineapps.com/news.php?rssid=1509 > > Thread here: > > [asterisk-users] University switches to Asterisk > > - -- > Cheers, > > Matt Riddell > ___ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://wap.sineapps.com (Daily Asterisk News for your cellphone) > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.2 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4 > KojTXtiqQLK7YNtXp+4Qh+I= > =T9d+ > -END PGP SIGNATURE- > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15 --Original Message Ends-- -- Melcon Moraes <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a gateway to SER
Siqhamo Sifo wrote: My asterisk is giving me problems when I use it as a pstn gateway to SER , basically what happens is that its either I get one way audio or no audio at all when I make pstn calls via asterisk from sip clients registered with SER. SER itself is just a SIP Proxy. So your issue may be the fact that you are not re-writing the SIP headers, if your endpoints are behind NAT. Diagnose the situation then provide detailed information, if you expect any assistance - We cannot read minds, yet. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dell hardware ...
Precisely our configuration. Dell 1850 with 4 port PRI digium cards. No issues on my last two consulting jobs. --On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon <[EMAIL PROTECTED]> wrote: Hi, Alan, We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards in it and it works perfect. It is almost Plug&Play. greetings ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dell hardware ...
This information seems to indicate there is a problem with the 1850 and the onboard nic. http://connection-telecom.com/support.html -Original Message- From: Ed Greenberg [mailto:[EMAIL PROTECTED] Sent: Saturday, September 16, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Dell hardware ... Precisely our configuration. Dell 1850 with 4 port PRI digium cards. No issues on my last two consulting jobs. --On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon <[EMAIL PROTECTED]> wrote: > Hi, Alan, > > We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards > in it and it works perfect. > It is almost Plug&Play. > > greetings ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.4/449 - Release Date: 9/15/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic
On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote: > I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel > : > > Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64 > x86_64 x86_64 GNU/Linux > > > and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s. > > I get this : > > laps1:~/Voipy/Bristuff/bristuff-0.3.0-PRE-1s/zaptel # modprobe zaphfc > FATAL: Error inserting zaphfc > (/lib/modules/2.6.13-15.11-smp/misc/zaphfc.ko): Invalid module format > > and this in dmesg : > > zaphfc: version magic '2.6.13-15.11-smp gcc-4.0' should be '2.6.13-15.11-smp > SMP gcc-4.0' This means you built zaptel with the wrong kernel headers. Is there a SUSE power user in the crowd? > > > What am I doing wrong? Anyone sucessfully using latest Bristuff under Suse ? A bit of trivia: kapejod (the developer of bristuff) is known to be using SUSE as the main system. Not sure of the version. However this is basically a kernel module building question. As such you can also ask in in more specific SUSE places. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Reliability of the newer IAXy's
> -Original Message- > From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] > Sent: Friday, September 15, 2006 4:37 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non- > Commercial Discussion > Subject: Re: [asterisk-users] Reliability of the newer IAXy's > > This goes for everyone; > > With an Asterisk server directly assigned a public static IP and with > the client SIP devices behind a NAT (usually Linksys routers) I have > yet to have a NAT issue, even when the client is behind dual NATs. > Of which MY issue is that both the client and asterisk would be behind NAT boxes. And I was looking for something a little easier than trying to tyrick SIP into working. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or address (6)
On Fri, Sep 15, 2006 at 11:09:51AM -0500, Juan Miguel Yamakawa wrote: > Help me please.. > > ZT_SPANCONFIG failed on span 1: No such device or address (6) > > how can i fixed this problem. This means that a span linein /etc/zaptel.conf did not fit the spans that exist on your system now (see /proc/zaptel/* . Each file under /proc/zaptel represents a span). What card(s) do you have? What's the output of: cat /proc/zaptel/* To generate a configuration that matches your currently-loaded modules, use xpp/utils/genzaptelconf . Use the option -d to try to detect the cards on your system. Note that those are not necessarily the same cards you'll have next time system boots. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] SHSU asterisk installation?
yes in did, this could be one excellent case study for all asterisk Community! Please let me know when it will be held! By the way, keep asterisk community following your steps would be great for knowledge of everyone and to solve possible problems you could find! On 9/16/06, Melcon Moraes <[EMAIL PROTECTED]> wrote: That would be superb!-Original Message-From: Aaron Daniel < [EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial DiscussionCc:Sent: Sat, 16 Sep 2006 11:32:00 -0500 Delivered: Sat, 16 Sep 2006 13:21:48Subject:[asterisk-users] SHSU asterisk installation?Maybe I should just hold a conference call about all this stuff.On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote: > -BEGIN PGP SIGNED MESSAGE-> Hash: SHA1>> Roy Sigurd Karlsbakk wrote:> >>> Linked from /. today,> >>> http://www.networkworld.com/news/2006/091206-von-sam-houston.html> >>> talks about the Sam Houston State University (SHSU) migrating a> >>> rather large amount of users to asterisk. The article describes the > >>> installation in rather vague terms, so I was wondering if someone> >>> know how they plan to do this, in detail.> >>> >> Yes, there have been several threads about this. > >> > Obviously I _have_ tried to google about this, so if you could point me> > to one of those threads, I'd be grateful>> - From 3 days ago:>> http://www.sineapps.com/news.php?rssid=1509>> Thread here:>> [asterisk-users] University switches to Asterisk>> - --> Cheers,>> Matt Riddell> ___ >> http://www.sineapps.com/news.php (Daily Asterisk News - html)> http://wap.sineapps.com (Daily Asterisk News for your cellphone) > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss)> -BEGIN PGP SIGNATURE-> Version: GnuPG v1.4.2 (MingW32)> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org>> iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4> KojTXtiqQLK7YNtXp+4Qh+I=> =T9d+> -END PGP SIGNATURE- > ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra.Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15--Original Message EndsMelcon Moraes < [EMAIL PROTECTED]>___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read variable from shell script
On Sat, Sep 16, 2006 at 01:15:01PM +0200, Christophorus Laube wrote: > Hi list, > > is it possible to call a shell script from * which returns a number or a > string which can be read to an asterisk variable? Something like > 'Set(VAR(System(/opt/scripts/something.script)))? > Does anyone have an idea? http://pbxfreeware.org/app_backticks.c -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
So with that said, can anyone recommend a way that I can get a sip client on a cell phone that uses H.263 and amr to talk to an asterisk system? Is it just not possible because of licensing? It sounds kind of lame to have a sip client that can't talk to anything else because of codecs.. Steve Kennedy wrote: > On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: > > >> I have been searching, but I have not found the answer.. How might I add >> the amr codec to my asterisk server? >> I believe I found the amr source from >> http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip >> I compiled it but did not end up with any .so files like I thought I >> would need to put it into asterisk. >> Any pointers on how to get an amr codec into asterisk would be most >> helpful.. >> > > AMR is patent encumbered, just because the source is available doesn't > mean you can use it without a license. > > Voiceage (at least) run licensing for AMR. It's about $1 per license > (simulataneous encode or decode) with a minimum of something like $50K. > > > Steve > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: [Asterisk-video] VXIasterisk is available !
Yeh well I don't and I have just as much right to this list as you do. I think it's fine for a company, commercial or otherwise to post product announcements in the non -biz list. As long as they only post it once, and it's not an announcement about a price change or something equally lame (like this Friday 20% off polycoms etc). But if a company has gone to all of the effort to develop and asterisk add-on and they announce the launch in the users list once, then I'm grateful for the heads up. You guys act like this is hallowed turf or something. It's not, it's a user list and this product certainly interested me so thanks for the heads up. Anyone else, feel free to delete. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-video- > [EMAIL PROTECTED] On Behalf Of Michael.Benarouch > Sent: Tuesday, 15 July 2003 11:34 PM > To: Development discussion of video media support in Asterisk > Subject: Re: [Asterisk-video] VXIasterisk is available ! > > ya fully agree > Micke > - Original Message - > From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]> > To: "Development discussion of video media support in Asterisk" > <[EMAIL PROTECTED]> > Sent: Saturday, September 16, 2006 5:08 PM > Subject: Re: [Asterisk-video] VXIasterisk is available ! > > > > better post this sort of spam in -biz > > commercial products belong there, not here, or perhaps nowhere > > > > roy > > > > On 14. sep. 2006, at 11.59, [EMAIL PROTECTED] wrote: > > > >> After launching the FFasterisk project, "i6net" is launching VXIasterisk > >> : the > >> VoiceXML browser for Asterisk. > >> The VoiceXML browser supports Voice and Video with Asterisk. > >> > >> VXIasterisk allows to create easily powerful video portals for > >> Asterisk. > >> > >> For more informations, http://products.i6net.com > >> > >> > >> i6net > >> ___ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> asterisk-video mailing list > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-video > >> > > > > --- > > "Humans mostly aren't particularly evil. They just get carried away by > > new ideas, like dressing up in jackboots and shooting people, or dressing > > up in white sheets and lynching people, or dressing up in tie-dye jeans > > and playing guitars at people" > > - Terry Pratchett > > --- > > Roy Sigurd Karlsbakk > > [EMAIL PROTECTED] > > > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-video mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-video > > > > > > -- > > No virus found in this incoming message. > > Checked by AVG Free Edition. > > Version: 7.1.405 / Virus Database: 268.12.4/448 - Release Date: 14/09/2006 > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-video mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-video ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediatrix 1204
Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mediatrix 1204
Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calling to PSTN newbie question
Hi list, I have a asterisk and sopthones working well, and I make a configuration for it work with a voice gateway Addpac 2120 (4port FXO y 4 ports FXS), I have connected my gateway to my PBX, when I try to call to PSTN from my softphone, I have a trouble that Asterisk add the number 9 over the number that I dial. How can ignore that number 9 when I dial from my softphone? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling to PSTN newbie question
In your extensions.conf you probably have: exten => _9nxxnxx,1,Dial(ZAP/g1/${EXTEN},180,r) Change that to: exten => _9nxxnxx,1,Dial(ZAP/g1/${EXTEN:1},180,r) Of couse use the values that you need, not that exact example... On 9/16/06, Pablo Almido <[EMAIL PROTECTED]> wrote: Hi list, I have a asterisk and sopthones working well, and I make a configuration for it work with a voice gateway Addpac 2120 (4port FXO y 4 ports FXS), I have connected my gateway to my PBX, when I try to call to PSTN from my softphone, I have a trouble that Asterisk add the number 9 over the number that I dial. How can ignore that number 9 when I dial from my softphone? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On 16 Sep 2006, at 20:38, Net Nut wrote: So with that said, can anyone recommend a way that I can get a sip client on a cell phone that uses H.263 and amr to talk to an asterisk system? Is it just not possible because of licensing? It sounds kind of lame to have a sip client that can't talk to anything else because of codecs.. Well Asterisk does not _have_ to have an amr codec for you to be able to use your handset. If you have several of these handsets or other devices that support amr, then asterisk can route calls between them, just passing the stream through. If you want any of the interesting asterisk features, then it will need to transcode, and then Steve's right, not only do you have to add codec code to Asterisk (which is almost certainly a GPL violation) you also have to pay the patent holder for any commercial use of the codec. Your best hope is if a few of us can persuade Digium to support amr in the same way that they license g729 . Tim. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom programmable buttons
Hello list!Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see how you can "remap" a key function. Any help would be great!TIARon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call across 2 asterisks
So the config is stored in one db shared between two asterisk? What you need to do is setup an IAX trunk between the two machines and then tell it to dial UA2 on the local machine, and then the extension for UA2 on the 2nd asterisk. Setup both the same on the other machine. Now do you have the users always fixed to one machine? You would need to figure a solution for CDR and voicemail of not... CDR and voicemail you can do in a database as well.. On 9/15/06, unplug <[EMAIL PROTECTED]> wrote: How can I make this work (UA1 makes call to UA2)? If I issue a command Dial(SIP/UA2), the call will fail. Someone in the forum said that Asterisk doesn't support the configuration below. But I am thinking about the possibility to make it works. Anyone can share if it really works? UA1UA2 --NAT1NAT2 | | | | Asterisk1 Asterisk2 \ / \ / - (ARA) DB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: help connecting cell phone, chan_bluetooth
Mauricio Mantilla gmail.com> writes: > > > Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this guide: > http://www.thetechguide.com/howto/asterisk/chanbluetooth.html > Everything has worked ok, but when I actually want to start > asterisk, my phone doesn't connect all the way. All I'm getting in the > asterisk CLI is this:[AG] Motorola < AT+BRSF=23and my phone keeps asking the pin number in order to get paired with asterisk, but apparently it never gets validated. > I changed the hcid.conf file so it wouldn't ask for a pin > number, and now the phone doesn't ask for it anymore, but I think it > still can't connect, cause I'm still getting the same message in the > asterisk CLI, every few seconds.Does anyone have an idea of what could be wrong? can it be something wrong with my phone or my bluetooth dongle? or something about configuring asterisk?Any help would be really appreciated, > > > Mauricio Mantilla > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I am following the same guide using trixbox, but after I 'tar xzf bluetoothfiles.tar.gz' and then 'cd /usr/src/asterisk' and perform the 'make clean' I get the following error: make: *** No rule to make target `clean'. Stop. Any thoughts I why that may be? Can I use these same instructions for trixbox or do I have to go back to Asterisk version pre-trixbox? Thanks---Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Reliability of the newer IAXy's
On 2006-09-15 13:42:21 -0700, "Lists" <[EMAIL PROTECTED]> said: Not sure what to tell you. But for the price, I might have to try one of these instead: http://cgi.ebay.com/IAX-Native-FXS-for-Digium-Asterisk-VoIP-PBX-Beats-IAXy_W0QQitemZ130026839028QQihZ003QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem There is also the AG168V which is a relatively inexpensive ATA that can be configured for SIP or for IAX depending on firmware flash. I have a couple of these and they work well. I never got the IAXy because it is so limited with respect to codecs and kind of expensive also... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SHSU asterisk installation?
And I for one would like a transcript or recording of it! PaulH Melcon Moraes wrote: That would be superb! -Original Message- From: Aaron Daniel <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Sent: Sat, 16 Sep 2006 11:32:00 -0500 Delivered: Sat, 16 Sep 2006 13:21:48 Subject:[asterisk-users] SHSU asterisk installation? Maybe I should just hold a conference call about all this stuff. On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Roy Sigurd Karlsbakk wrote: Linked from /. today, http://www.networkworld.com/news/2006/091206-von-sam-houston.html talks about the Sam Houston State University (SHSU) migrating a rather large amount of users to asterisk. The article describes the installation in rather vague terms, so I was wondering if someone know how they plan to do this, in detail. Yes, there have been several threads about this. Obviously I _have_ tried to google about this, so if you could point me to one of those threads, I'd be grateful - From 3 days ago: http://www.sineapps.com/news.php?rssid=1509 Thread here: [asterisk-users] University switches to Asterisk - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFDCScS6d5vy0jeVcRAmQAAKCOcWU04FK6txwH1NfduA0QsFYaogCfb3a4 KojTXtiqQLK7YNtXp+4Qh+I= =T9d+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1158424231.497848.25931.almora.hst.terra.com.br,5369,Des15,Des15 --Original Message Ends-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send DTMF down a channel
Hi, Actually I have several AGI scripts, manager "Originates" and Call files doing different stuff applying the same procedure, thats why I asked about console messages. It seems very weird to me. Regards On 9/16/06, Frank Church <[EMAIL PROTECTED]> wrote: I followed my intuition and sent a DTMF tone via the D() option in the dial command to the destination and that causes the voice to be transmitted after the call is answered. I also realised that the ringing tone to the destination does not come up until I execute a SendDTMF command before dialling the destination. >From that it appears that a command to issue a tone either before dialling out or after the call is picked up is necessary for the exchange of sound to take place, and I don't know whether that is a bug, or a feature of the Asterisk design. In verbose mode the call is interspersed with the output from the complicated AGI script and that will make things difficult, I have the problem (hopefully) solved now, but I will try to make the time to send you the output of the script after I remove the AGI code from it. In dial plan terms the idea is to generate a call file that contains both the caller id of the caller and the destination, and when the caller answers the call is routed to an AGI that extracts the dialled number and dials it as though they were entered via DTMF. And it appears that because no sound stream is generated through entry of DTMF, no sound is transferred after the call is answered. Perhaps trying something like that in a dial plan you create yourself will reproduce the problem. Thanks Frank On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote: > could you post the output of the asterisk console in verbose mode? > > In logger.conf > > [logfiles] > console => notice,warning,error,verbose,debug > > Regards > > On 9/15/06, Frank Church <[EMAIL PROTECTED]> wrote: > > The program in question is an adaptation an AGI calling card program. > > It is adapted for callback by setting by channelling the callback call > > into the context used for the normal inbound leg. > > > > When used that way entering the PIN and destination number via DTMF > > works normally. > > > > I tried to implement sms/web callback simply by passing the > > destination number and the callerid in call file's parameters, so that > > the script can go straight to dialling the destination number without > > prompting the caller for it. > > > > This method results in dead air when the call is answered at the > > destination and it seems that the absence of DTMF tones for generating > > the call is the cause. > > > > Logically it should work, why the call should result in dead air > > because the destination number was not obtained through DTMF is what I > > can't understand. > > > > It appears that the absence of DTMF stops the RTP from being > > established to the destination number or prevents the calls from > > being joined. When I dial back to the calling number I get the busy > > voice prompt okay. > > > > Injecting the DTMF tones on the call channel is a really a kludge to > > simulate actual key presses in the hope that the dead air problem will > > go away and it . > > > > On 9/15/06, Moises Silva <[EMAIL PROTECTED]> wrote: > > > Frank. PlayDTMF and SendDTMF is the same as pressing keys at the > > > phone. Im not understanding well, can you please explain a practical > > > scenario of how do you expect it to work, and how actually works? :) > > > Thanks > > > > > > Regards > > > > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote: > > > > How do you specify the channel the SendDTMF is sent on, as it has to > > > > be variable? I am able to use PlayDTMF to send tones using the > > > > manager interface - the tones come up okay but it doesn't work. > > > > > > > > My problem comes from a working callback script that I am trying to > > > > adapt to web/SMS callback. > > > > > > > > The script is unchanged only that in the web/sms callback the > > > > destination number is passed in the call file variables to allow it be > > > > dialled after the callback is answered rather than being obtained via > > > > DTMF. > > > > > > > > The destination call is made alright, but the called party only gets > > > > dead air after the call is answered. It seems that the fact of > > > > pressing the DTMF keys activates the sound channel in way that the > > > > direct call back does not, which is why I am going through all these > > > > hoops. I just hoped that sending some DTMF tones programmatically > > > > would simulate the actual key presses. > > > > > > > > Do you have any ideas of what the problem might be? > > > > > > > > > > > > > > > > On 9/14/06, Moises Silva <[EMAIL PROTECTED]> wrote: > > > > > http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF > > > > > > > > > > Regards > > > > > > > > > > On 9/14/06, Frank Church <[EMAIL PROTECTED]> wrote: > > > > > > How can DTMF be sent down a channel? > > > > > > > > > > > > I am thinking of method where say
[asterisk-users] Wrong outgoing port
Hi, I have just compiled trion*CLI> show version Asterisk 1.2.12.1 built by root @ trion on a i686 running Linux on 2006-09-16 16:39:13 UTC I have copyed the old configuration files from the stable debian version 1:1.0.7.df I have changed in sip.conf the bindport (port=) to (bindport=) . When i try to connect to my provider it doesn't work becourse it try's to connect to port 5060. sip debug says Retransmitting #5 (NAT) to 62.177.135.42:5060: REGISTER sip:62.177.135.42 SIP/2.0 trion*CLI> sip show registry 62.177.135.41:38383 31##120 Request Sent What's wrong. Somehow it looks like a bug. Regards, M. Piscaer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mediatrix 1204
I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp <[EMAIL PROTECTED]> wrote: Bill Michaelson wrote: > Would anyone be kind enough to post a sip.conf fragment as a sample for > use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
Steve Kennedy wrote: On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. While the Voiceage licence for G.729 seems to cover you for patent issues on that codec, their licence for AMR does not. Their patent pool does not include all those who claim a patent on AMR. So, you need their licence plus you need to search out and licence other people's patents too. I assume this is why the AMR licence appears cheaper than the G.729 one. Steve' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call across 2 asterisks
On 9/17/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: So the config is stored in one db shared between two asterisk? Yes, the configuration files are shared between them. What you need to do is setup an IAX trunk between the two machines and then tell it to dial UA2 on the local machine, and then the extension for UA2 on the 2nd asterisk. Setup both the same on the other machine. Do you mean my configuration works? Could you express it a little bit more? On 9/15/06, unplug <[EMAIL PROTECTED]> wrote: > How can I make this work (UA1 makes call to UA2)? > If I issue a command Dial(SIP/UA2), the call will fail. Someone in > the forum said that Asterisk doesn't support the configuration below. > But I am thinking about the possibility to make it works. Anyone can > share if it really works? > > UA1UA2 > --NAT1NAT2 > | | > | | > Asterisk1 Asterisk2 > \ / >\ / > - (ARA) DB > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] system cmd
How can I use a system cmd to get back the return value in dial plan? Say, I want to run a script using system cmd to get the hostname. System(hostname) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users