similar to: Configuring Asterisk to allow any payload type in SDP

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488

Hi All. I have an application where the sample rate of the sender and receiver can vary by a small margin and the latency needs to be maintained within bounds and can't drift significantly and the system has to be able to cope with clock mismatches up to 0.5%. For example, the sender may have a clock rate of 48.1kHz and the receiver may have a clock rate of 47.9kHz. Unfortunately the clock

The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. 11.02.2016 1:47,

Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value "a=sendrecv" is present, according to the rfc3264 the sdp value a must be mark with "sendonly". I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and

Hello, I have added the following to the peer definition : ignorecryptolifetime=yes But still Asterisk tells me : [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32 [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process: SRTP crypto

Kia ora I'm a using lme (from nlme package) with data similar to the Orthodont dataset and am getting positive log-likelihoods (>100). This seems usual and I wondered if someone could offer a possible explanation. I can supply a sample dataset if requested, but I feel almost certain that this question has been asked and answered recently. However, I can find no trace of it in the mail

Kia Ora Let's say we have: "myreplacementfunction<-" = function (..., value) { call = sys.call () print (as.list (call) ) 0 } Then we call: x = 0 myreplacementfunction (x, y, z) = 0 It will return: [[1]] `myreplacementfunction<-` [[2]] `*tmp*` [[3]] y [[4]] z $value <promise: 0x06fb6968> There's two problems here. Firstly, x has to be defined otherwise we

Dear Sir, I current have done successfully with sip message over asterisk server , and additionally now I want to send the image between sip using asterisk. Could any one share me how to config the asterisk for sending image from sip? Thank, I am waiting for your reply. Thyda -------------- next part -------------- An HTML attachment was scrubbed... URL:

Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Kia Ora everybody. There must be an obvious answer to this, but I can't see it.... I want four square plots in one postscript file. The canonical answer would be: postscript(file="~/f.ps",width=5,height=5) par(pty="s",mfrow=c(2,2)) plot(1:19,xlab="") plot(1:19,xlab="") plot(1:19,xlab="") plot(1:19,xlab="") dev.off() But this

On Tue, 2024-02-27 at 16:46 +1300, June Chong | TechnologyWise via samba wrote: > Hi team, > Is there a way to grab Kerberos specific log entries? > Example: > /Auth: [Kerberos KDC,ENC-TS Pre-authentication] user.../ > I have tried using the kerberos class but nothing was logged when I > specified a path. > This is what I have on my smb.conf. > /[global] log level =

Dear All, Apologies for this simple question and thanks in advance for any help given. I want to make from my .R script an .exe file. Is there any way to transfort my script to an autolaunch file? It means it runs the script by double clicking on it. p.s.: I'm using windows -- View this message in context: http://www.nabble.com/executable-script-tf4806651.html#a13751752 Sent from the R

Hi I need to set the number of incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is: cat sip.conf [general] [template](!) qualify=yes cc_agent_policy=generic cc_monitor_policy=generic call-limit=2 busylevel=1 callcounter=yes subscribecontext = hint allowsubscribe=yes [100](template)

<!DOCTYPE html> <html><head> <meta charset="UTF-8"> </head><body><p>Hello list,</p><p><br></p><p>I am asking when a caller want to leave a message to a mailbox with the application voicemail</p><p>How i can limit the duration for exemple 30 seconds.</p><p>exten =>

I mean by sending image by using sip channel just like we can send text message and what about sending image file ? On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp <jcolp at digium.com> wrote: > On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote: > > Dear Sir, > > Kia ora, > > > > > I current have done successfully with sip message over asterisk server , > >

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Kia Ora > Although I'm not sure what problem it would solve... Given that you asked, I was interested in writing a multiple assignment function as a replacement function, so something like: massign (x, y, z) = construct.some list () Obviously, that's not possible. Probably the best example I can think of is converting cartesian coordinates to polar coordinates. Then we might have

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

Hi Jean, This is my last attempt at trying to use speex for a project we are working on. I have contacted you before, but i have had no joy what so ever trying to find a working example in c++ that shows how to add a header to a raw speex file, so that it an be played using windows. If you do know of a working example can you please supply a link. As this is a commercial project maybe you could