similar to: Help_In Voicemail , vedio play but voice is not here out.

Yes, DAHDI is a timing source and meetme depends on DAHDI for voice mixing. You can check details here http://www.asterisk.org/docs/asterisk/trunk/applications/meetme >Install DAHDI then !!? >On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra < >durgesh.mishra at rancoretech.com> wrote: >> In make menuselect =>application=>XXX app_meetme . I am doing confrence >>

Thanks, as per my knowledge, after downloading the source code, and making changes and then making sure it works properly , recompiling it and then patch it and finally submit a pull request. But what all changes should I/can I make? Please provide me some insight into this. Thanks&Regards, Hemant Kumar Singh On Apr 10 2019, at 10:56 am, Gaurav Arora <gauravarora.daiict at gmail.com>

Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information:

Okay thanks, how do I make a patch and submit one? Do you have a link for the same? Thanks&Regards, Hemant Kumar Singh On Apr 10 2019, at 4:13 am, Olly Betts <olly at survex.com> wrote: > On Tue, Apr 09, 2019 at 04:07:58PM +0530, Hemant Kumar Singh wrote: > > The GSoC guide section recommends reading of the hacking. > > > I don't actually see a reference to

Hi, Can anyone suggest me how to remove the uploaded images.I am uploading images using file column.I want to remove the images both from the database as well as from the folder where i am saving the images. Thanks&Regards, Chandra -- Posted via http://www.ruby-forum.com/.

Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India.

Dear I have a data.frame, and want to fit a constrained non-linear model: data: x y -0.08 20.815 -0.065 19.8128 -0.05 19.1824 -0.03 18.7346 -0.015 18.3129 0.015 18.0269 0.03 18.4715 0.05 18.9517 0.065 19.4184 0.08 20.146 0 18.2947 model: y~exp(a)*(x-m)^4+exp(b)*(x-m)^2+const I try to use nls() and set start=list(a=1,b=1,c=1,m=1), but which always give me a error message that

Hi Folks, I am just trying to get started up in RoR, I have done a simple application of add, edit, delete.... I am now trying to accomplish pagination in RoR, I referred a few tutorials, however none of the examples that i tried from there, seemed to work error free..... I have heard that lots of things have deprecated in RoR, can someone please post me a detailed report of how i can

HI, Getting below error while trying to compile openssh-5.8p2 on *Centos 5.6_X86-64* *configure: error: *** Can't find recent OpenSSL libcrypto (see config.log for details) **** I recently compiled latest open ssl version *OpenSSL 1.0.0d 8 Feb 2011 *please help me to solve this issue. ** Thanks&Regards, Abdul Jabbar * *

hi, Am a software developer having 4 yr experience in c++.I want to integrate R environment in my c++ application,please help me to do so. thanks&regards blesson [[alternative HTML version deleted]]

Hi all, I have the following questions regarding Xen hypervisor.Can you please clarify these queries? Does Xen follow any security model, in particular, does a Random Oracle (RO) fit in Xen? When there are concurrent Guest OS running on the same hardware, then there has to be a mechanism for concurrency control and fairness, how does Xen implement these? Shared memory access has to make sure that

Hello all and thanks, I have encoded several .vob with " ffmpeg2theoar -p preveiw " with no problems at all. The quality is exellent. These .vob were ripped with vobcopy. The problem I am having is when I encode a .vob with " ffmpeg2theora -p pro ". The resulting .ogg will not play correctly with vlc or kaffeine. The vedio is jerky and the sound go out of sycro just

Hello list, I'm trying to set up a video call from my Ekiga client to a Grandstream GXV3140 IP-phone. The call succeeds but there is no video. I have in sip.conf : videosupport=yes disallow=all allow=alaw allow=g726 allow=g729 allow=gsm allow=h261 allow=h263 allow=h263p allow=h264 The Grandstream peer has codecs (sip.conf) : gsm;alaw;g729;h261;h263;h263p;h264 The Ekiga peer has codecs

Hi all, I realise that asterisk's codec negotiation has been discussed in the past multiple times. What I haven't been able to understand is how asterisk decides which video codecs to advertise to the other end when canreinvite=no in sip.conf and the initial caller doesn't support video. My tests are quite simple, I use an asterisk with 4 peers all on the same LAN. My sip.conf

Hi, we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42

Hi, The only reason I use Samba is I want to connect my linux desktop with my windows laptop and share the larger disk with the laptop. I got weird problem that I could only write to the samba server but I could not read from it with errors like: [2001/03/04 16:36:38, 0] lib/util_sock.c:write_socket_data(540) write_socket_data: write failure. Error = Broken pipe [2001/03/04 16:36:38, 6]

Hi, I tried to turn on Video and get the following cli-WARNING output -- Executing [8600 at outgoing-kamailio:1] Answer("PJSIP/7000-00000000", "") in new stack > 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 -- Executing [8600 at outgoing-kamailio:2] ConfBridge("PJSIP/7000-00000000", "8600") in new stack --

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

Hi all Maybe somebody has an idea. I'm tracing a very strange phenomena... I've a connection from Asterisk to a SIP PBX. Most calls have a caller ID. Some International calls don't have any. Now it looks like those calls without caller ID never get to the context where incomming calls from this SIP PBX should get to.... Examples: Call with Caller ID: (slightly anonymized)