search for: phonea

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 situati...

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 S...

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 a...

Hi all, I have a problem with using remote MySQL database server with two 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 A...

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...

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

...C REFER is send from Phone B to proxy: U 192.168.10.124:5060 -> 192.168.10.75:5060 REFER sip:180 at 192.168.10.75:25060 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.124:5060;branch=z9hG4bK_7C2F8008F5C1_T5B47A547;rport. From: <sip:phone-b at 192.168.10.75>;tag=7C2F8008F5C1_T1484940980. To: "PhoneA" <sip:180 at 192.168.10.75:25060>;tag=as18f69e58. Call-ID: 452af6610540b7cf0d4c49f372d46779 at 192.168.10.75. CSeq: 2 REFER. Refer-To: <sip:200 at 192.168.10.75?Replaces=CALL_ID9_7C2F8008F5C1_T1871060912% 40192_168_10_124%3Bto-tag%3Das13cb6557%3Bfrom-tag% 3D7C2F8008F5C1_T570909484>...

...e the Polycom "reject" action but through the Asterisk dialplan. Reasons: 1. It would allow me to log through Asterisk who's rejecting calls 2. It would allow rejecting calls from another phone (see above scenario) I thought there could be a "SendSIPCode 486 to SIP peer phoneA" application, but a quick scan of the documentation does not bring obvious answers. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180508/7709d359/attachment.html>

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

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 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...

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

...weird problem, it seems to happen at random periods throughout the day from a few minuets to a up to an hour. [Phone A] >--SIP--> [Asterisk] >--SIP--> [Phone B] Both phones are snom 360's. Asterisk is Stable 1.0.9 Pretty simple config, just a dial direct to each other like Dial(SIP/phoneA,30,t) Running Gentoo linux When we make a call during one of the problem periods, from [Phone A] extension to [Phone B] there is up to 2 seconds delay before A hears B. Looking at a packet trace we see the SIP invites coming in and the calls being setup ok, but on the call to B we see the RTP fro...