similar to: Examples of Asterisk deployments with 100-500 users?

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

Hi- Having a little trouble with the SMS application in the UK. I've read the wiki topic and also the BT documentation SIN 413, which describes the service technically. I've been able to successfully send messages from my customer's ISDN landline to various mobiles, but cannot get the mobiles to be able to send messages back to the landline. I've tried sending from both Cellnet

I was wondering if anyone has put together a comprehensive list (that is reasonably maintained) that lists country codes, landline numbers, mobile numbers, etc. The particular requirement is for a dialplan to know what is going to be charged to whom. For example, mobile and landline rates are the same in the US the US has a unified numbering plan of 1NXXNXXXXXX, while the UK has: 441xxx

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Hello R users! I like to use cat in a loop. I know, loops are not the best way in R ... but my question: It is possible to overwrite the expression "Reading row: i" in each iteration of the loop (print out in the below loop on the screen) or more particulary to overwrite the counter "i". for(i in 1:header$M){ cat("Reading row: ", i) SparseIndex[[i]] <-

Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.

I''m having a bit a trouble getting some search terms to work with acts_as_ferret. Particulary where multiple words are in the query. For instance if I search for "Time to Kill" it doesn''t find the any records. However, if I search just for time, Time to Kill comes up. I''m not sure what I''m missing here. Any help would be greatly appreciated. --

Hi, I have a question for you: Let’s suppose I have three physical servers; two on the same area, and the last one elsewhere (not on the same network, but all three have connectivity between each others). May I use this infrastructure with libvirt? I mean in terms of cloud automation: is it possible to remotely deploy VMs (with particulary configurations) with libvirt? For example, is it

Hello Newbie question here: I have a box running Ubuntu Linux 7.10 "gutsy gibbon", and have a single Digium TDM410E card, with 1 FXO module fitted and connected to my landline. I have it answering the landline, directing to SIP phones, diverting to voicemail etc - and it works great. What I can't work out is how to dial Out from this single card. It is possible? if so, is

Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using