search for: timotsmith

Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated in a RAD modem and they give me ethernet to the E1 card. It's the first time i am having install such a solution, which ideally would be not a big problem. However, The physical layer has failed to

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have...

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have...

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

------ Original Message ------ From: "Tony Kasule" <timotsmith at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: 30/06/2015 8:34:47 p.m. Subject: Re: [asterisk-users] Help With Physical Layer >Hello, > >Anyone to help me with this issue? It has never worked :( H...

Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello users, > > I have a Digium Te235 and asterisk 13 which have worked well with 1 > carrier but we have failed to add a 2nd carrier. The second telco brings > their E1 line over finer, terminated in a RAD modem and they give me > ethernet to the E1 c...

Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make

Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this

On Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull <duncan at e-simple.co.nz> wrote: > > Hi Tony > > I'm not familiar with the card you but 120 ohm is usually twisted pair, > and 75 ohm is coax (usually). If it is changeable its usually done with > jumpers on the card. > The new Digium cards have no jumpers anymore and I don't think they support coaxial cables.

Hi, I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user but Sip user hears. - Calling the PSTN

Hi, Could someone please help me with this? I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below. - For conversations between analog phone and sip phone, Analog phone can't here the SIP user

Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no success. i have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and =/usr/sbin/ast-playlinein

Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and other tutorials on the net but no success. I have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and

Hi, In our office, we're migrating from a Cisco set up to Asterisk. We'd like to do it gradually, so I've added an asterisk server as an H.323 gateway to the call manager so out going calls are going through asterisk. So far so good. Am now faced with the challenge relaying incoming calls from asterisk to call manager. Has anyone done that before? I won't be allowed to just make

Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the queue. My question is, how can it be possible for call to skip other calls in the queue and be picked

Hi, I am running a service where I play full songs but MP3 files kept on crashing my server. I resorted to wav but the quality is really poor after converting..or even sometimes not audible at all! Do you guys know of a better way I can convert mp3 to wav and restore quality? Below is the script I am using, I also tried the steps at

Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Its the way some files are encoded. Is there a way I can make it skip the files that can be played? I use the

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an