search for: ast_verb

...===================================================================== partial code of app_confbridge.c: ==================================================================== static const char *const app2 ="MyConfbridgeCount"; static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); ast_verb(3 ,"\n ==Inside load_module==\n "); ast_log(LOG_NOTICE ,"\n ==Inside load_module==\n "); //tes4 //const char *data= (char*)malloc(sizeof(char) * 256); char *sdata="400...

...EFINE(count_exec, "Show Number of User(s) in Conference." ), AST_CLI_DEFINE(admincount_exec, "Show Number of adminUser(s) in Conference." ), }; ==================================== /*! \brief Called when module is being loaded */ static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); int res = 0; //static const char * const app = "ConfBridge"; //static const char * const app = "ConfBridge"; if (conf_load_config(0)) { ast_verb(3, "Unable to load config. Not loadi...

...const char *data) { int res = 0; struct conference_bridge *conf=NULL; int count; char *localdata; char val[80] = "0"; struct ao2_iterator i; //struct conference_bridge *bridge = NULL; struct conference_bridge tmp; AST_DECLARE_APP_ARGS(args, AST_APP_ARG(confno); AST_APP_ARG(varname); ); ast_verb(3,"\n============Inside count_exec =============\n"); if ( ast_strlen_zero(data)) { ast_log(LOG_WARNING, "MyConfbrigeCount requires an argument (conference number)\n"); ast_verb(3, "\n MyConfbrigeCount requires an argument (conference number\n "); return -1; } if...

...tls_session->parent->tls_cfg->ssl_ctx)) ) { SSL_set_fd(tcptls_session->ssl, tcptls_session->fd); if ((ret = ssl_setup(tcptls_session->ssl)) <= 0) { ssl_err=SSL_get_error(tcptls_session->ssl,ret); ast_verb(2, "Problem setting up ssl connection:ssl_err=%d, %s\n", ssl_err,ERR_error_string(ERR_get_error(), err)); if(ssl_err==SSL_ERROR_SYSCALL) { if( ret == -1) ast_verb(2, "Problem setting up...

...*********** *** 626,635 **** --- 630,650 ---- watchers[pos++] = in; for (o = outgoing; o; o = o->next) { /* Keep track of important channels */ + if (ast_test_flag64(o, OPT_SINGLE_BUSY)) + ast_verb(2, "OPT_SINGLE_BUSY set\n"); /* always set, why? */ if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan) watchers[pos++] = o->chan; numlines++; } + /* I'd like t...

...const char *data) { int res = 0; struct conference_bridge *conf=NULL; int count; char *localdata; char val[80] = "0"; struct ao2_iterator i; //struct conference_bridge *bridge = NULL; struct conference_bridge tmp; AST_DECLARE_APP_ARGS(args, AST_APP_ARG(confno); AST_APP_ARG(varname); ); ast_verb(3,"\n============Inside count_exec =============\n"); if ( ast_strlen_zero(data)) { ast_log(LOG_WARNING, "MyConfbrigeCount requires an argument (conference number)\n"); ast_verb(3, "\n MyConfbrigeCount requires an argument (conference number\n "); return -1; } if...

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: # grep C-00024cd5 full.log | egrep 'Strict RTP' [Feb 22 11:16:41] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c: > 0x2b308c074f80 -- Strict RTP learning after remote address set t...

...hank you for that. From the code it kind of looks like > STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: > > if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) > && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), > rtp->rtp_source_learn.start)) { > ast_verb(4, "%p -- Strict RTP learning complete - Locking on source > address %s\n", > > Our call shows: > > # grep C-00024cd5 full.log | egrep 'Strict RTP' > [Feb 22 11:16:41] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c: > > 0x2b308c074f80 -- Strict RTP le...

Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension.

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum