I just realize in these few days there are many calls that already hangup but not detected by Asterisk. Those calls occupy PSTN lines and need to be manually terminated through Flash Operation Panel or phycally disconnect the PSTN lines. This never hap..
Im looking for a method to setup FoIP i.e. using T.38 protocol with no PSTN lines. I tested T.38 feature for Asterisk but the problem Im getting is unable to send more than 2 pages but getting timeout error. Past couple of years I also configured ..
Im looking for someone who can help us setup Fax with T. 38 on asterisk 10.x.x – We need to be able to do FoIP (Fax over IP) as we have no pstn lines available. Do you know how to setup a reliable fax system, then we will pay you to help us do th..
Tim, Thanks for your response. Here is my topology as listing down below; PSTN Line –> Cisco Voice GW –> IP Cloud –> Asterisk Will Asterisk able to receive the fax (as in topology above) using its fax module? In sip.confI enabled fax detection ..
all, Couple of things I would like ask, does Asterisk provides free license for FoIP (for 1 channel) or need to purchase it? Couple of years back, I was able to send and receive the fax using Digium T1 card, in term of FoIP how can I able to rece..
list, We are interested in configuring an Asterisk based GSM server in Africa an I would like your recommendations. The requirements are: 1) A system with at least 8 GSM cards 2) 2 pstn lines from here (the US) will directly link with 2 lines on ..
I have a server with an OpenVox A400P card with 2 FXO modules on it. The internal extensions are SIP Grandstream phones. When making or receiving external calls through PSTN, there is an interrupted hissing like high pitch noise – which might go a..
, I am from SaudiArabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me toresolve this is..
1-Check signaling type on gateway PSTN ports 2-Set RTP timeout in SIP trunk. From: firstname.lastname@example.org [mailto:email@example.com] On Behalf Of Mike Sent: Friday, March 04, 2011 7:46 PM To: Asterisk Users Mail..
I think I have 3 PSTN lines because I can connect a normal telephone to them all and make calls between each of them. We have 5 normal telephones and 1 panasonic. From what I got I need a PCand a of PCI card to interface to my 3 external lines and..