We, along with a lot of other people, have a phone number that is pretty important to us. Yesterday, our VoIP provider went down… wont call any names VI, but it was pretty bad…Our goal is to create a script within asterisk, that will place a c..
On several occassions lately, my home Asterisk box has stopped registering with my VoIP provider.I havent been able to reproduce the problem, and the log doesnt contain anything useful.How can I increase the log verbosity for SIP registration-rela..
I m looking for a way to pass the 302 moved temporarily received from the SIP device back to the SIP provider. Here is the setup:Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done ..
List,Since Im looking for a new VoIP provider for US origination/termination, Iwill very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.msThanks in advance!Elder D. Arohuanca dCAP 1497L..
I work for a VoIP provider in Southern California. We are looking for someone very knowledgeable in Asterisk/VoIP to help work on the following: – Maintenance of current Asterisk servers, updating Asterisk, monitoring load, and other sysadmin task..
I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected dynamically. Wanted to check with the community if this feature holds true on latest versions of Asterisk ? Regards, Mitul Limbani, Chief Architech & Founder, Enterux Soluti..
I am having a problem with SendDTMF()- 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(wwwww3wwwww2wwwwww1wwwww4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 w..
If you are providing a hosted phone system to customers, how do you deliver the calls? If you are a end user, how does your provider deliver the calls to you? The reason I ask is I read and hear a lot of issues where people are getting dropped ca..
On Mon, Jul 18, 2011 at 03:20:03PM +0200, Gilles wrote: > > > Id like to run Asterisk on an embedded device, where space is scarce. > It should be able to handle calls from a VoIP provider in SIP, calls > from the PSTN through Dahdi, and voicemail..