similar to: Video doesn't work for outgoing call?

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

Hello, i'm tring to use a cox's model for a survival analysis. I have a dataset, this is a part: VOD SESSO fonte_sct donor RT_CGY STATOBMT TEMPO morto 1 0 F mid related 1200 CP 65 1 2 0 M mid 1200 2RC 5281 0 3 0 M mid unrelated 1200

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

hello i was wondering what is the use of "rinstance" in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case "rinstance" parameter is missing. when i dial "SIP/username" audio is fine but when i dial SIP/x.x.x.x:port there is one way audion.

Does this help: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0 Method: REGISTER Request-URI: sip:1.2.3.4;transport=TCP Request-URI Host Part: 1.2.3.4 [Resent Packet: False] Message Header Via: SIP/2.0/TCP 192.168.1.15:47053 ;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP

Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says "Scheduling destruction of sip dialog xxxx ". Then it says "Really destroying sip dialog xxxx". What to do for this problem??? I

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

Hello, I'm having difficulty with registering a SIP account in a Snom 320 IP-phone. This is what sip debug tells me : [Oct 7 13:28:42] VERBOSE[20314] chan_sip.c: [Oct 7 13:28:42] <--- SIP read from UDP:public_ip:58697 ---> REGISTER sip:sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.114.200:2048;branch=z9hG4bK-vj1xvbdnp4dw;rport From: <sip:test3 at

Dear vod vos, On 2020-10-04 6:47 p.m., vod vos via R-help wrote: > Hi, > > a <- c(1, 4) > b <- c(5, 8) > > a:b > > [1] 1 2 3 4 5 > Warning messages: > 1: In a:b : numerical expression has 2 elements: only the first used > 2: In a:b : numerical expression has 2 elements: only the first used > > how to get: > > c(1:5, 4:8) The simplest way is

Look for nginx-rtmp Should do the job perfectly.. ?????? ??? ??, 2 ????? 2017 ?-16:37 ??? Nitin N <nitin.workz at gmail.com>: > Dear All, > > I am curious to know if IceCast can be used for audio/video on-demand > streaming over http/https. > > My basic understanding is that IceCast runs in broadcast mode. As such, it > may not be ideal for my particular use case,

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379

libswfdec/swfdec_player.c | 18 +-- test/trace/loadvars-decode-5.swf |binary test/trace/loadvars-decode-5.swf.trace | 182 +++++++++++++++++++++++++++++++++ test/trace/loadvars-decode-6.swf |binary test/trace/loadvars-decode-6.swf.trace | 182 +++++++++++++++++++++++++++++++++ test/trace/loadvars-decode-7.swf |binary test/trace/loadvars-decode-7.swf.trace |

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Dear All, I am curious to know if IceCast can be used for audio/video on-demand streaming over http/https. My basic understanding is that IceCast runs in broadcast mode. As such, it may not be ideal for my particular use case, wherein, I wish to stream media files on demand to multiple browser based clients who may request the same media file at different times. Is it possible at all then to

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hello List, following this thread: http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains I tried to configure on the pjsip.conf the same endpoint with different domains like: [1000 at sip.domain.com] type=endpoint [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console:

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Hi, i'm trying to apply StepAIC with coxph...but i have the same error: stepAIC(fitBMT) Start: AIC=327.77 Surv(TEMPO,morto==1) ˜ VOD + SESSO + ETA + ........ Error in dropterm.default(fit,scope$drop, scale=scale,trace=max(0, : number of rows in use has changed: remove missing values? anybody know this error?? Thanks. Michele [[alternative HTML version deleted]]