search for: phonec

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured somthing correctly or is there a bug?? Later. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Ou...

Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA --> asterisk --> phoneB phoneA (music on hold), phoneB --attended call transfer--> phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause t...

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server Error".

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | | | |INVITE B | | | | |----...

...Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA (which is registered with Asterisk#1) from PhoneC (which is registered with Asterisk#2) and call fails because "cause 3 - No route to destination". This is the record (among other data) for PhoneA in MySQL: fullcontact/ipaddr/port/regserver sip:361 at 10.2.4.15:5060;user=phone;transport=udp/10.2.4.15/5060/asterisk "sip show peers&...