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  r377594 | igorg | 2012-12-10 00:56:04 -0600 (Mon, 10 Dec 2012) | 15 lines
  
  Fix codec mismatch
  
  Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. 
  
  (issue ASTERISK-20183)
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  Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 377592 from http://svn.asterisk.org/svn/asterisk/branches/10
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  Merged revisions 377593 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r377595 | igorg | 2012-12-10 01:03:48 -0600 (Mon, 10 Dec 2012) | 3 lines
  
  Add firmware information to CLI devices listing
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@377602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Automerge script
2012-12-10 07:19:34 +00:00
parent db0e7049ac
commit 58f0ade4db
1 changed files with 9 additions and 6 deletions
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15
channels/chan_unistim.c
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@@ -446,6 +446,7 @@ static struct unistimsession {
int size_buff_entry; /*!< size of the buffer used to enter datas */
char buff_entry[16]; /*!< Buffer for temporary datas */
char macaddr[18]; /*!< mac adress of the phone (not always available) */
char firmware[8]; /*!< firmware of the phone (not always available) */
struct wsabuf wsabufsend[MAX_BUF_NUMBER]; /*!< Size of each paquet stored in the buffer array & pointer to this buffer */
unsigned char buf[MAX_BUF_NUMBER][MAX_BUF_SIZE]; /*!< Buffer array used to keep the lastest non-acked paquets */
struct unistim_device *device;
@@ -2666,9 +2667,9 @@ static void send_start_rtp(struct unistim_subchannel *sub)
buffsend[16] = (htons(sin.sin_port) & 0x00ff);
buffsend[20] = (us.sin_port & 0xff00) >> 8;
buffsend[19] = (us.sin_port & 0x00ff);
buffsend[11] = codec;
}
buffsend[12] = codec;
buffsend[11] = codec; /* rx */
buffsend[12] = codec; /* tx */
send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_tx), buffsend, pte);
if (unistimdebug) {
@@ -2697,9 +2698,9 @@ static void send_start_rtp(struct unistim_subchannel *sub)
buffsend[16] = (htons(sin.sin_port) & 0x00ff);
buffsend[20] = (us.sin_port & 0xff00) >> 8;
buffsend[19] = (us.sin_port & 0x00ff);
buffsend[12] = codec;
}
buffsend[11] = codec;
buffsend[11] = codec; /* rx */
buffsend[12] = codec; /* tx */
send_client(SIZE_HEADER + sizeof(packet_send_open_audio_stream_rx), buffsend, pte);
} else {
uint16_t rtcpsin_port = htons(us.sin_port) + 1; /* RTCP port is RTP + 1 */
@@ -4320,6 +4321,7 @@ static void process_request(int size, unsigned char *buf, struct unistimsession
if (unistimdebug) {
ast_verb(0, "Got the firmware version : '%s'\n", buf + 13);
}
ast_copy_string(pte->firmware, (char *) (buf + 13), sizeof(pte->firmware));
init_phone_step2(pte);
return;
}
@@ -5932,12 +5934,13 @@ static char *unistim_show_devices(struct ast_cli_entry *e, int cmd, struct ast_c
if (a->argc != e->args)
return CLI_SHOWUSAGE;
ast_cli(a->fd, "%-20.20s %-20.20s %-15.15s %s\n", "Name/username", "MAC", "Host", "Status");
ast_cli(a->fd, "%-20.20s %-20.20s %-15.15s %-15.15s %s\n", "Name/username", "MAC", "Host", "Firmware", "Status");
ast_mutex_lock(&devicelock);
while (device) {
ast_cli(a->fd, "%-20.20s %-20.20s %-15.15s %s\n",
ast_cli(a->fd, "%-20.20s %-20.20s %-15.15s %-15.15s %s\n",
device->name, device->id,
(!device->session) ? "(Unspecified)" : ast_inet_ntoa(device->session->sin.sin_addr),
(!device->session) ? "(Unspecified)" : device->session->firmware,
(!device->session) ? "UNKNOWN" : "OK");
device = device->next;
}