similar to: SER + Asterisk

Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:

Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also

I recently posted a message about an interesting dilemma which I could not figure out. After doing a whole lot more digging I think I have figured out part of what is going on. The parking app is implemented as an app, ie can be called from extensions.conf) but if you look really closely, it is also implemented in each channels driver (handler, whatever you want to call it) and those

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at

Is there a way to ring selective line on multi-line phone. For example if I'm on the phone talking internally on line 1 and the calls comes-in the line 2 will automatically ring. The phone P104 allow extension to be assign each line. Is there a way to call certain line (example line 3) on multi-line phone instead of line 1 when the phone is not busy? For example the Sip phone P104 has

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Sorry if this is the wrong list... I need a toll-free number to be delivered to me on IAX. (This is NOT an existing number need to buy the whole service.) Anyone know of a service provider offering this? -Mark 707-735-1038

If I have different opening hours on different days, can I accomodate that in a single gotoiftime, or will I need to filter them out one by one ? For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00 can this be done something like GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or something like that, or do I have to do:

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

Muchas gracias Carlos !! un abrazo, Eric. On 30-05-19 17:57, Carlos Ortega wrote: > Hola, > > Sí, en ggplot esto se resuelve con: > > * facet_grid() > * facet_wrap() > > Saludos, > Carlos Ortega > www.qualityexcellence.es <http://www.qualityexcellence.es> > > El jue., 30 may. 2019 a las 23:30, neo (<ericconchamunoz en gmail.com >

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Dear ALL: I find a program named "asterisk_b2bua" on http://developer.berlios.de/projects/b2bua/ And I also download them(two components) and try to test it. But I have not enough knowledge about asterisk. It seems a Software PBX. Does asterisk_b2bua work? Does anybody ever try it? I have questions about my scenario. |======================> UA2

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Dear All, I have the following setup. Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; 1. How do I

Hi all, I would like to know if it is possible to use asterisk and ser together in a single computer system using ser as a sip proxy and forwarding any voice call request to asterisk for calling into the pstn gateway. (or any other alternative that is possible is also welcomed for suggestions). If it is possible can someone kindly show me the necessary configuration files or refer me to any page

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?