search for: ubiquitysoftware

...E1 line? As always thanks for any insights / advice people can give. cheers alex (My background is over 4yrs in SIP so if you have SIP protocol questions feel free to ask) -------------------------------------------- |Alex Barnes |SQA Engineer |URL : http://www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> -------------------------------------------- This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited....

...biquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net <http://www.ubiquity.net/> and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ub...

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

...biquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net <http://www.ubiquity.net/> and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ub...

...risk PBX so am definitely to be considered a newbie. cheers Alex (I also have a few Pingtel hardphones, Grandstream Budgetone 100, Snom 100 somewhere and a broken Cisco 7960) -------------------------------------------- |Alex Barnes |SQA Engineer |URL : http://www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> -------------------------------------------- This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited....

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.

Hi all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here >> http://www.g7ltt.com/VoIP/vmfiles.html <http://www.g7ltt.com/VoIP/vmfiles.html> And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc)

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

-----Original Message----- From: Peter Svensson [mailto:psvasterisk@psv.nu] Sent: 25 November 2004 10:54 >> They work just fine if your pstn provider is at all serious. If not, >> switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during power outage I was simply speculating on the

> -----Original Message----- > From: Ronald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through

> -----Original Message----- > From: Michiel van Baak [mailto:michiel@vanbaak.info] > Sent: 26 May 2005 20:22 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > > > Anton, > > My script is not connecting to the manager interface. > The php script is run as agi script as first when a call > comes in. The

> > > >| -----Original Message----- > >| From: asterisk-users-bounces@lists.digium.com > >| [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >| Michiel van Baak > >| Sent: Martes, 24 de Mayo de 2005 04:45 p.m. > >| To: asterisk-users@lists.digium.com > >| Subject: Re: [Asterisk-Users] CallerID > >| > >| On 23:44, Tue 24

Hi all, Hopefully you can help me. I want to turn off the audio "Call Waiting" beep that plays during a call. I have found the line in the indications.conf for Call Waiting but apart from setting the frequency to zero or the length to zero is there a proper way to disable this functionality. thanks very much alex This email and any attached files are confidential and

> -----Original Message----- > From: David Liu [mailto:david@deltapath.com] > Sent: 31 January 2005 14:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Announcement to caller when > called party haspicked up - without initial Answer()? > > > This is super easy to do. All you need to do is to put that > announcement

Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk