similar to: Pb on data transmission size

Hi all, I am just coming in the mailing list.... ;-) I'm working on an Ripper/Encoder for GNUstep. At first sight, libvorbisogg seems to be the library that I needed. The problem for me is documentation. So I have a few question about this library : - in order to init an encoding process, we need to provide channels. Well, what does is mean ? - in order to init an encoding process, we need

Hi all, I am just coming in the mailing list.... ;-) I'm working on an Ripper/Encoder for GNUstep. At first sight, libvorbisogg seems to be the library that I needed. The problem for me is documentation. So I have a few question about this library : - in order to init an encoding process, we need to provide channels. Well, what does is mean ? - in order to init an encoding process, we need

Hi all ! I use cdparanaia for which I would like to get information. Does anyone know if there is a mailing list concerning this ripping application. In fact, I would like to know if cdparanoia can support scsi drive .... if someone can help me on this matter ... it is welcome. ;-) <p>Thanks for your help ! <p>Ce message contient des informations confidentielles ou appartenant

Hi,all When we allocate bandwidth to a host/device in LAN , how can we know the max. bandwidth the device(e.g.1394,ethernet,powerline..) can receive(in the idle condition, no other application occupy the bandwidth) ? Can RSVP signaling do that?or others ? Thanks Fionna ___________________________________________________ 最新版 Yahoo!奇摩即時通訊 7.0，免費網路電話任你打！ http://messenger.yahoo.com.tw/

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Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

Hello listamtes, Here's is a curious issue: netcat (nc) seems to not do UDP for me, even though TCP works flawlessly. Not even on the local host. Here's the "server session" I am running: cat test.dd | nc -l -u -n -k -v 2456 (test.dd is just a large file with random data) Here's the client: nc -v -u 127.0.0.1 2456 | dd of=/dev/null ^C0+0 records in 0+0 records out 0

Hi Folks, I used to run transmission for torrents on opensolaris snv_134. My working zfs was newsan/tmp and when the torrent completed i had transmission configured to move the files to newsan/incoming. I upgraded to sol11exp and the related zfs updates and now the files no longer move to newsan/incoming upon completion. I believe this to be a zfs acl issue and could use some assistance in

Dear friends - a couple of papers in PNAS (lastly:framework for making better predictions by directly estimating variables' predictivity, Lo et al PNAS 2016; 113:14277-14282) have focused interest on mapping complex traits to multiple loci spread all over the genome. I have been around on the relevant taskview(s) I hope but fail to see that the backward haplotype transmission association

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I'm having an intermittent problem with fax transmissions, something I call the streaking mascara syndrome. TxFax will be part way through a transmission when something fails--on the fax printer, part of the page will have correctly printed, but the rest of the page is "streaked" to the bottom. I have several successful multi-page transmissions interspersed with streaked

Hello everybody, I'm experiencing a strange problem with my Asterisk. I hope you can help: Asterisk is running at my company behind NAT. Ports 5060 and 10000-20000 are being forwarded to it. I have put the router's external IP-address into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN 7050 which is registered with the Asterisk at my company. When I try to call

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

I'm trying to use the txfax application based on spandsp in Asterisk 1.2. It seems to be working but I would need a way to reliably check whether the fax has been completely transferred or not. I'm using a mail2fax system (as with email2fax and .call files) but I can't seem to get it working. If I use "Application" and "Data" in the .call file, there doesn't

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Hi, I just installed Antonio Gallo's agx-ast-addons package in order to use app_txfax with asterisk-1.4. Compiling according to docs went well. However, I'm getting an error after the first page of fax: /usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec: Transmission loop error The (very first) page is transferred perfect anyway. Then app_txfax unfortunetly stops the transmission.

Build with qt5 (from epel) & devtoolset-2 Sources: ------------ transmission-2.82-1.el6.src.rpm https://drive.google.com/file/d/0B9RlkKQB1POScXlWU3M5cE5QVW8/edit?usp=sharing libevent2-2.0.21-1.el6.src.rpm https://drive.google.com/file/d/0B9RlkKQB1POSbUNPQmNoTDRXdmM/edit?usp=sharing Binaries: ----------- https://drive.google.com/file/d/0B9RlkKQB1POSQ2Q2T3dHbEt3enM/edit?usp=sharing

Wayne, I've been ruminating a bit on the qsorted flist and duplicate removal and have an idea. What if we removed end_file_entry() from send_file_name() and instead had a for(i=0; i < flist->count; ++i) send_file_entry(flist->files[i], f, ??) in send_file_list(). As near as i can tell this would be unnoticeable to the protocol. Once done the sort of the file list could be