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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...

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My situation is that I have...

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...

...pdate connected line info on ringing phone C. Request is sent directly from asterisk to phone: U 192.168.10.75:25060 -> 192.168.10.102:2048 UPDATE sip:phone-c at 192.168.10.102:2048;line=jrtagx14 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.75:25060;branch=z9hG4bK7bc63ea0. Max-Forwards: 70. From: "PhoneB" <sip:100 at 192.168.10.75:25060>;tag=as2b910e05. To: <sip:phone-c at 192.168.10.75>;tag=dg06weh6iz. Contact: <sip:100 at 192.168.10.75:25060>. Call-ID: 2dadc6b61efcb4dc726be746564897bf at 192.168.10.75. CSeq: 103 UPDATE. User-Agent: IPTAM PBX (Version 20141216/6814). P-Asser...

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 can't seem to retrieve a parked call. Here is what I do: Call from phoneA to phoneB Answer phoneB Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park th...

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...

...172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server 172.20.2.5: Phone A-->asterisk A----->SER----->asterisk B--->PhoneB All devices all have ip connectivity (No Firewalls! No Natting) to each other. so phone a can ping phone b and server b, etc, etc, etc.. Can reinvite is enabled on both the ser connection (on both sides) and for both phones.. Making a call from phone A to phone B works great.. Except you can he...

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 | | | |...

I have an ISDN line with 10 numbers. The line is then connected to * with one HFC-based card. The format of the numbers is like below: 123456-0 123456-1 ... 123456-9 Now I would like to connect those numbers to different telephones, i.e. when someone dials 123456-0, he/she is connected to the digital receptionist. If someone dials 123456-2, the connection goes to SIP/202 If someone dials

This is a real 'newbie' type question, but I can't get my brain to work today. Is it possible to re-direct an incoming SIP call based on it's CLI? Ideally I would like to check incoming calls against a short whitelist (of say 20 numbers) and redirect to a different extension if there is a match. I would also like to redirect calls that fail to present any CLI (aka